For at least 14 years after the publication of minimum permissible exposure limits that would largely eradicate industrial deafness, statute legislation in Queensland remained unchanged and ineffective. Industrial deafness continued to occur. New legislation, introduced in 1989 and amended in 1993, and based on a duty of care responsibility incumbent on all, may remedy this situation. The new legislation is examined and comments are made about the values inherent in the new approach. It is concluded that public policy strategists may increase the likelihood of success of they ensure that the duty of care provisions (together with the general provisions of the Act) are backed up by innovative complementary economic, financial and marketing incentives.

Recent studies reveal that grip forces due to repeated mechanical vasocompressions are most significant for the genesis of vibration-induced which finger syndrome (VWF). Therefore, exerted grip force was regarded as a dependent variable in 2 experiments and the effects of noise and vibrations of different weighted acceleration levels were studied. Neither grip forces nor peripheral blood flow as indicated by finger skin temperature were influenced by noise or vibrations. the cause of VWF is therefore presumed to be a concomitant variable which correlates with weighted accelerations and with grip forces as well. A possible factor is the weight of hand-held vibrating tools.

The persence of distinct tones in environmental noises leads to an increasing anaoyance especially if the tones are of long duration. Before 1992 in Germany the tonal content of noises had to be subjectively estimated by the acoustic consultants and up to six dB could be added to the measured Leq depending on the tonality content of the noise under consideration. In order to give an objective basis for tonality estimated a DIN norm proposal for tonality evaluations was introduced in 1992. This proposal is compared with two different procedures : the prominece-ratio calculation which was proposed by Bienvenue and Nobile /4/ and the proposal of Ares /5/. The output of the three models is contrasted with subjective tonality judgements. It turns out that the prominence ratio model yields the best agreement with the subjective tonality assessments od the sound set chosen.

The impacts of road traffic noise in Hong Kong are pervasive. About one million peoople are affected by road traffic noise at levels higher than a standardd recommended fro planning of new developments. The Environmental Protection Department of Hong Kong has promulgated a set of planning standards and guidelines for reference of planners, engineers and architects in their preparation of land use proposals which include road and residential developments. This paper will describe, in connection with road traffic noise in Hong Kong, the planning objectives, the various practicable mitigation measures available to a high density modern city, and the achievements through conscientious planning efforts made over the past years.

This paper presents the design and construction details of a soundproof enclosure for housing 20 KVA diesel generator-set. As the generator had to be installed close to the hospital building, it was desirable to reduce the transmission of noise by housing the generator in such an enclosure. The diesel engine being an air cooled one, it was essential to supply fresh air into the enclosure for its cooling. Forced inflow of air is provided through an inlet duct located in such a way that the incoming fresh air is thrown close to the inlet of cooling fan of the engine. The high velocity air stream, which heats up while passing over the engine head, escapes to the atmosphere through a rectangular outlet duct with enlarges inlet that receives hot air from the engine. The air ducts were designed specially and have been provided with acoustic lining for sound absorption. The masonary enclosure has been provided with double glazed fixed windows and double doors. The exhaust pipe of the engine fitted with a muffler has been taken out through the enclosure wall facing away from the hospital. Acoustic performance studies conducted in terms of attenuation provided by the enclosure at different frequencies have also been presented and discussed. The noise control measures adopted for building the sound-proof enclosure have been found to be quite effective as the noise levels inside the hospital building are now within the acceptable limits.

The paper presents a simple method for calculating the sound exposure level (LAE) of helicopter noise. It is assumed that a helicopter is a nondirective point source and that A-weighted sound pressure level at an observation point can be expressed by an A-weighted power level and a simple function of the distance from the helicopter. We derived a formula for LAE by integrating the sound energy along a finite or an infinite flight segment. The values calculated form the formula agree well with the results of test flights in which three types of helicopters each were operated in three moving modes of approach, takeoff and level flyover.

Acoustic signal such as speech and scattered sound, are generally a nonstationary process whose frequency contents vary at any instant of time. For time-varying signal, whether a nonstationary or a deterministic transient signal, a traditional frequency domain representation does not reveal the contents of signal characteristics and may lead to erroneous results such as the loss of desired characteristics features or the mis-interpretation for a wrong conclusion. A time-frequency domain representation is needed to characterize such signatures. Pseudo Wigner-Ville distribution (PWVD) is ideally suited for portraying nonstationary signal time-frequency domain and carried out by adapting the fast Fourier transform algorithm. In this paper, the important properties of PWVD were investigated using both stationary and nonstationry signatures by numerical examples PWVD was applied to acoustic sigtnatures to demonstrate its application for time-ferquency domain analysis.

The acoustic phenomena in the actual sound systems involve a variety of compound problems. In this paper, the well-known Bayes' theorem is first employed and expanded into orthonormal and non-orthonomal series forms matched to the digital processing of lower and higher order statistical informations and the noisy observations. Proposed on-line algorithms of digital filter type are applied to the actual state estimation for a reverberation characteristics in a room under contamination of background noises.

For evaluating the response fluctuation of the actual environmental acoustic system excited by arbitrary random inputs, it is important to predict a whole probability distribution form closely connected with evaluation indexes Lx, Leq and so on. In this paper, a new type evaluation method is proposed by introducing three functional models matched to the prediction of the response probability distribution from a problem-oriented viewpoint. Because of the positive variable of the sound intensity, the response probability density function can be reasonably expressed theoretically by a statistical Laguerre expansion series form. The relationship between input and output is described by the regression relationship between the distribution parameters(containing expansion coefficients of this expression) and the stochastic input. These regression functions are expressed in terms of the orthogonal series expansion and their parameters are determined based on the least-squares error criterion and the measure of statistical independency.

In the actual sound environmental systems, it seems to be essentially difficult to exactly evaluate a whole probability distribution form of its response fluctuation, owing to various types of natural, social and human factors. Up to now, we very often reported two kinds of unified probability density expressions in the standard expansion from of Hermite and Laguerre type orthonormal series to generally evaluate non-Gaussian, non-linear correlation and/or non-stationary properties of the fluctuation phenomenon. However, in the real sound environment, there still remain many actual problems on the necessity of improving the above two standard type probability expressions for practical use. In this paper, first, a central point is focused on how to find a new probabilistic theory of practically evaluating the variety and complexity of the actual random fluctuations, especially through introducing some equivalence transformation toward two standard probability density expressions mentioned above in the expansion from of Hermite and Laguerre type orthonormal series. Then, the effectiveness of the proposed theory has been confirmed experimentally too by applying it to the actual problems on the response probability evaluation of various sound insulation systems in an acoustic room.

In the actual sound environment, the random signal often shows a complex fluctuation pattern apart from a standard Gaussian distribution. In this study, an evaluation method for the sound environmnetal system is proposed in the generalized form applicable to the actual stochastic phenomena, by introducing two type information processing methods based on the regression model of expansion series type and the Fuzzy probability. The effectiveness of the proposed method are confirmed experimentally too by applying it to the observed data in the actual noise environment.

The insertion loss is the measured change in power flux at a specified receiver, when the acoustic transmission path between it and the source is modified by the insertion of silencer element. Such measurements have clear and valid physical meaning particularly if the source impedance remains while the transmission path is altered. When the invarient condition is satisfied, the insertion loss is given by the ratio of the acoustic pressure in upstream to that in downstream of the silencer, and that of the particle velocity. The measurement is consisted of using an adaptation of the two microphone method to obtain the complex amplitude of the sound in upstream tube as well as in downstream tube of the silencer. Examples of the data, reduced and presented in terms of the pressure ratio and particle speed ratio, are compared with the theoretical calculations.

The program name TOLAPS is an acronym for Take-Off LAnding Profile Simulation. Some of the interesting features of this program is the ability to detect flight performance effects of airport altitude, ambient temperature, air pressure and wind. TOLAPS can also handle effects of TOW and LW. The program user can also calculate profiles by user difined flaps and thrust settings deviating from recommended standard settings for each aircraft. Wind effects on straight out flying as well as turns can also be demonstrated. Output form TOLAPS are either screen graphics of profiles (altitude, speed or thrust versus flight distance) or flight track. Profiles can also be made in a tabular form, ready for use in most airport noise calculation programs. In this way, TOLAPS is a valuable tool to evaluate effects of noise abatement procedures.

The goal of this study is to develop treatment methods for improving the noise quality of Home Appliance Drain Pumps. Developed treatment methods covers the various methods including the control of the water level through the developed electronic circuity and the various modifications over the design of impeller and fan units. In the first part of the studies Dominant noise sources of the pumps are analyzed. Specific noise problems of the pumps are classified on the basis of mechanic and magnetic origins. Factors including the pressure variations at suction head, magnetic interactions with structure, specific fan noises are studied sequentially. The second part of the studies are considered for development and application of treatment methods. Results denoted the basic problems and ways to improve the noise quality by treating the dominant soureces of drain pumps.

In two previous papers [1], [2], we presented the validity of a method that calculated the Leq values along High Speed Train (TGV in french) lines from the level/time evolution of moving trains. Tanks to this method, it is now possible to compute specific time-related effects such as interactions between train bodies and close obstacles. This paper lists important parameters to be considered within TGV studies and presents the various levels of study, starting from the research of the best traject (extensive studies), passing through noise impact studies (intensive studies) of the chosen traject to the dimensionning of antinoise devices (final design), and all this to guarantee precise respect of noise criteria. A theoretical comparison study conducted on about 80 different types of antinoise devices including earthberms and noise barriers of different forms, dimensions and materials is also presented. At last a "final design" study using all benefits of the method (full 3D and time representation) is presented.presented.

Scientific foundation for ultrasonic scale preventing devices construction was given in 40S-50S of this century but their production in former USSR was organized later in 70th. Several different principles of scale preventing is overviewed together with physical principles of ultrasonic method. Practical experience received in USSR in 80S is discussed. Technical decisions and inventions used for construction of the first device UZU-1 produced in Cheboksary plant are enumerated and principles of UZU-2 device are briefly sketched.

Elastic wave propagation in discrete random medium is studied to evaluate the effects of particle resonance on dispersion and attenuation of composite materials containing spherical inclusions. The frequency-dependent wave speed and attenuation coefficient can be obtained from proposed self-consistent method. It can be observed that the abrupt increase of effective wave speed and the concurrent peak of attenuation at low frequency is due to the lowest resonance of particles, whereas those in high frequency region are due to higher ones. The lowest resonance is mainly caused by the density mismatch and higher resonances by the stiffness mismatch between matrix and particles. The dispersion and attenuation of elastic waves in particulate composites are affected by the lowest resonance much than by higher ones.

In the actual situation of measuring the environmental noise, it is very often that only the resultant phenomenon fluctuation contaminated by the additional noise of arbitrary distribution type can be observed. Furthermore, the observed data is usually given in a sound level form the purpose of estimating only the undisturbed objective output response, some estimation method is necessary to reasonably remove the effect of the above additional noise. In this paper, first, a mathematical model of arbitrary sound insulation systems is introduced in the form of a linear system on intensity scale, by using the well-known additive property of energy quantities. Next, some estimation method of the output response under the existence of background noise is derived. Then, based on the expression of the above estimation method, a new prediction method of only the output response probability function form for arbitrary sound insulation systems without. a background noise is proposed by use of observed data contaminated by a background noise. Finally, the effectiveness of the proposed method is confirmed experimentally too by applying it to the actual various type sound wall systems.

The four-terminal transmission matrix method has been widely used to estimate the insertion-loss. However, the predictins using the equations in the four-terminal transmission matrix method do not reflect a practical phenomenon accurately, In this paper, the correction method to derive the insertion-loss for a constant sound pressure source is presented. The method of correction to the four-terminal transmission matrix method was proposed by rewriting the real and imaginary parts as they depend solely on the flow velocity. Then the result was compensated for by adding the component of the temperature gradient.

Very often, one would like to have visual image of mechanical or acoustical events such as musical sound and transient vibrations. Conventional methods to visualize the signal, such as power spectrum, do not normally allow to cultivate the signal of interests due to their inherent limitation on transient signals. Other than the conventional method, one could use an instantaneous frequency which can reveal the variation of frequency in terms of time. Nevertheless it is quite sensitive to noise and can not resolve the frequency components of signals; normally produces additional components other than those of the signals. In this paper, we introduce the Wigner-Ville spectrum to see the transient characteristics of signal, especially musical sound and transient mechanical vibration signatures. For musical sound, several popular western classic music have been selected for the analysis. For the transient mechanical signature, the signals obtained from the car door experiment and the beam experiment are interpreted in terms of Wigner-Ville spectrum. Results demonstrate the visual expressions of transient signals; musical sound and vibrations.

The work discusses the problems of modelling of the process of acoustic signal generation in machines. We have pointed out that in the task of minimizing of both moise and vibration, the key problem is identification of sources and paths of propagation, both in terms of their location and of definition of their characteristic features. Properly conducted identification makes possible the use of relatively simple mathematical models and this fact is particularly important for a broad application of the proposed methods in practice.

The impedance of an electro-acoustic transducer can be controlled by motional feedback, and the noise in a duct can be reduced actively by adjusting the impedance using an additional sound. In this paper, two approaches for active noise control using motional feedback (MFB) loudspeaker are described. First configuration uses an external sensor to pickup of source directly. In this configuration, the adaptation of controller is necessary to compensate the change of transfer function from noise source to control poing. The second configuration uses a new adaptive algorithm specialized for peridic noise. Because this configuration does not require any reference input and the error sensor couples very tightly with control loudspeaker, this MFB system itself is independent of the duct condition. No microphone are required in both configurations, so that a more reliable and stable active control system can be realized under severe conditions such as high pressure, high temperature, dust, flow and so on.

The volume of an acoustic source is important in determining various acoustic parameters. One of the suggested techniques is the internal pressure method incorporating a loudspeaker attached to a chamber wall and a microphone inserted into the cavity. Although the method is easy to handle with a very simple measurement setup, the coupling effects between the dynamic system of the loudspeaker and acoustic field, and the effects of higher order modes introduced by the discontinuities in the acoustic field, and the effects of higher order modes introduced by the discontinuities in the acoustic field should be considered for precise result. In this study, higher order modes due to the discontinuities of loudspeaker and microphone boundaries are included and the electro-acoustic coupling effects are compensated for by using the results of two cylinders with different lengths. The volume velocity of a loudspeaker thus obtained agrees very with that measured by laser sensor.

A new simple method to formulate the adaptive algorithm to control the coefficients of FIR filter is introduced. The filter is used in the active noise control system. The introduced algorithm includes the LMS algorithm as a special case. The validity of the theoretical result is confirmed by the computer simulation.

Detecting partial discharge(PD) and locating its source are one of many diagnosis methods. Location the PD source is very important to reduce the time and cost of repairing power transformers. And to locate the PD source, the cross-correlation method is a well known one. But there many spurious peaks in cross-correlation, and occasionally, some peaks could be bigger than the true one. In order to analysis these spurious peaks and to reduce them, we have done many experiments and simulations. As the results we could reduce the spurious peaks, and get well defined cross-correlation from which it is easy to locate the PD source accurately.

The active noise control which regards the acoustic power as a target function to be minimized, is analyzed to test its feasibility of which simplifies the measurement system compared with the global acoustic energy based active noise control system. In fact, it is found that the acoustic power based active noise control strategy is equally likely as good as the global acoustic energy based active noise control method if the acoustic field of interest is diffusive or very low model density one. In the intermediate model density field, we also demonstrate that the power based control gives the similar results as the energy based control in terms of global sound energy reduction for the lightly damped enclosure which might be most important system in practical application. From all the theoretical and power based control strategy is dependent on the characteristics of the acoustic field to be controlled; i.e., the model density distribution, the degree of reverberation, and on the strength of modal interaction of the control source with the primary source; i.e., the location of control source.

In this paper, the echo phenomena of Yingying Pagoda(ancient Chinese architecture), which may be resulted from interferences of reflection and diffraction by the pagoda eaves when pulse sound source is at some suitable positions, are investigated by an 1:2 scale model. There are valleys in frequency spectrum due to the interferences. On the other hand, taking eaves as wedges approximately, numerical spectral estimates are obtained from the closed-form impulse solution for diffraction of pulse point-source radiation by an infinite rigid wedge. The results of the numerical computations are similar to those of the model experiments. The study is a helpful guide to reconstruction or maintenance of this kind of ancient buildings.

Measuring noise, sound quality or acoustical comfort presents a difficult task for the acoustic engineer. Sound and noise are ultimately jugded by human beings acting as analysers. Regulations for determining noise levels are based on A-weighted SPL measurement performed with only one microphone. This method of measurement is usually specified when determining whether the ear can be physically damaged. Such a simple measurement procedure is not able to determine annoyance of sound events or sound quality in general. For some years investigations with binaural measurement analysis technique have shown new possibilities for the objective determination of sound quality. By using Artificial Head technology /1/, /2/ in conjunction with psychoacoustic evaluation algorithms - and taking into account binaural signal processing of human hearing, considerable progress regarding the analysis of sounds has been made. Because sound events often arise in a complex way, direct conclusions about components subjectively judged to be annoying with regard to their causes and transmission paths, can be drawn in a limited way only. A new procedure, complementing binaural measurement technology combined with mulit-channel measuements of acceleration sensor signals has been developed. This involves correlating signals influencing sound quality, analyzed by means of human hearing, with signals form different acceleration sensors fixed at different positions of the sound source. Now it is possible to recognize the source and the transmission way of those signals which have an influence on the annoyance of sound.

There are strong ocean wave interference with big amplitude very low frenqencies are similar to the ship's hydrodynamic signals. To detect ship's hydrodynamic field will be faced various natural hydrodynamic interferences which are radom and the prior knowlege of which are not know. This paper proposes to use the adaptive noise cancelling principle and used the model of adaptive wave canceller to eliminate the ocean wave interfrence and detect the ship's hydrodynamic signals. Computer simulation results shown that signal to noise ratio can be raised from several to ten times. It shows the fact that this mathod can detect the ship's hydrodynamic signals from the strong ocean wave interferences while it is difficult for the old methods.

Two kinds of static and dynamic state estimation methods are newly discussed for the problem of the measurement disturbance of environmental low-frequency noise in the presence of wind-induced noise. First, the probability characteristics of wind-induced noise are discussed in the form of probability distribution conditioned by wind speed, based on the simultaneous observation of the wind-induced noise and wind speed near a microphone. Next, especially form the viewpoint of simplicity for practical use, two kinds of static and dynamic state estimation methods are discussed. The static estimation method using the information on wind speed is fundamentally supported by the conservation principle of energy sum. The dynamic one is the method by using a recursive digital filter with the parameters successively renewed by the information on wind speed. This can be also simplified by using well-know Kalman filter under the assumption of the Gaussian distribution. The effectiveness of proposed two estimation methods are shown through experiments under a breezy condition in the open filed.

Gardermoen is chosen as the location for a new major airport for the Oslo area, The site is surrounded by various units and camps operated by the Norwegian national defence. A study was carried out to evaluate whether the occurrence of aircraft noise may result in the national defence having to restrict operations in established camps, and in areas where outdoor exercise, training and instruction are beeing carried out.

High-frequency and complex vibration ultrasonic wire bonding systems are propsed and their welding characteristic are studied. Ultrasonic wire bonding is used widely for joining thin connecting wire of various electronic devices including IC or LSI. Conventional bonding systems use vibration frequency of 40 or 60 kHz and linear vibration welding tips. Complex vibration welding tip which vibrates in elliptical to circular or rectangular to square in the same or different frequency is effective to join welding specimens in shorter welding time and under smaller vibration amplitude, and furthermore high-frequency systems such as 90, 120, 190 kHz are also significantly effective. High-frequency and complex vibration welding system of 90, 120 and 190 kHz are designed. Welding characteristics of these systems are found very superior than a conventional system. Welding specimens of aluminum wire of 0.1mm diameter are successfully.

In this paper, a new method of statistically evaluating an output response probability distribution of a memory type non-linear system is practically derived based on a zero-memory type non-linear equivalent system. That is, first, the objective system is approximately and functionally separated into two functional parts, i.e., a zero-memory type non-linear part and a memory type linear part according to the well-known Wiener's idea. A whole mathematical frame of the output probability distribution is evaluated in an approximate but generalized form, based on the equivalent zero-memory type non-linear part. The memory effects between the input and the output of the system are reflected in the statistical parameters and the expansion coefficients.

This paper describes the global vibro-acoustic troubleshooting approach, used to identify and separate different sources of noise and vibrations on a boilerfeed pump testrig. The pump serves for rotor dynamic research of a EC-funded BRITE-Euram profect. This approach resulted in the identification of local structural flexibilities in the connections between the machinery and the base plate. The relative importance of the modes during normal operation is revealed by comparison with operational deformation shapes. The use of sound intensity mapping allowed to calculate the total sound power and to rank the equipment according to its sound power contribution. High acoustic levels were found and related to the fluid drive and to the piping system. Modification of the piping section resulted in a reduction of noise and vibration levels along the test loop and smooth operation in a wide suction pressure range.

This paper presents the nearfield acoustic holography (NAH) imaging system for airborne sound constructed in our own laboratory. The effects of different kinds of noise and the filter function's form in wave number space for reconstructions are analysed emphatically. Using the system we have measured the vibrating mode and the radiated field of a simulative "Chime Stone" made by metal, and gotten interesting results. These results indicate that NAH can be used to research the mechanism of sound production and evalution of musical quality for Chime Stone as an effective means.

This paper describes the concept and basic technique of measuring torsional operational deflection shapes using a laser-based torsional vibration meter, a dual-channel FFT analyzer and operational deflection shapes software running on a PC. Torsional Operational Deflection Shapes (TODS) is defined similar to ODS (Operational Deflection Shapes), with the exception the TODS designates the operational deflection shapes of structures vibrating in a rotational, or angular, degree of freedom. Thus the TODS measurements can be applied to rotating shafts and the results of such a measurement are shown. In some cases it may be great benefit to apply order tracking and/or synchronous time domain averaging techniques in order to avoid smearing and reduce noise problems.

As powertrain noise is better and better controlled, road noise inputs become more important. The interior road noise of a car is mainly induced by the wheels rolling over the road surface. Each of the four wheels act as an independent and uncorrelated excitation input. To rank the energy transfer form each input to the interior, a Transfer Path Analysis (TPA) needs to be made-which requires operational vibration measurements. However due to the multiple uncorrelated inputs, phase relations vary continuously. It is therefore necessary to separate the operational data into set of "independent phenomena" by means of a Principal Component Analysis (PCA). A TPA can then be carried out for each independent phenomenon. Operational deflection shapes referenced to these principal components share the physical phenomena. The details of the methodology are discussed and a discussion of the results on a car shows that the method gives accurate results for full vehicle testing.e testing.

Main sources of increased vibrations and air noise on ship are main and auxiliary engines and ship ducts. The various ways of transfer of vibration energy and air noise in passenger cabin of a vessel require, in general case, of various methods of attenuation. The transfer of vibration energy from engines through a support requires, alongside with shock-absorbers, availability active shock-absorbers. The transfer of vibration energy and hydrodynamic noise on ship ducts requires availability, alongside with flexible muffler, active mufflers. The availability of air noise from working equipment can require, along with absorbent covers, of space systems of active noise control. In the given article it is spoken about the unified approach to formation of the block-diagram of active noise and vibration control. The complex approach permits to receive additional efficiency in reduction of noise in passenger cabin of vessels.

The transfer function of an acoustic system, in general, often exhibits a wide dynamic range and a very long impulse response. The time-stretched pulse (TSP) proposed by Aoshima (ATSP) has a small peak-factor and is accordingly suitable for the measuring impulse responses. The pulse is not so suitable, however, for the measurement of impulse responses over a wide frequency range. In this paper, we try to generalize and optimize this method (OATSP). This makes the method applicable for measuring of impulse responses longer than the length of the TSP. An analysis of error in such a case is also shown. Finally, we discuss how to implement this technique in specific measurement conditins.

Application of the filtered-x LMS adaptive filter to active noise cancellation requires to estimate the transfer characteristics between the output and the error signal of the adaptive canceller. We analyze the effects of estimation accuracy on the convergence behavior of the canceller when the input noise is modeled as a sinusoid.

This paper reports on several experiments carried out to find the effects of intensity difference on duration discrimination in pairs of pure tones. Pure-tone signals were presented with the modification of amplitude and duration. Duration-discrimination thresholds for the musical tones and the signals that were varied in amplitude were investigated together with those of the fixed signals. From the results, it was found that the duration discrimination tasks are affected by an intensity difference of 20 phons.

This paper presents the Korean text-to-speech (TTS) algorithm with speed and intonation control capability, and describes the development of the Voice message delivery system employing this TTS algorithm. This system allows the Interpersonal Messaging (IPM) Service users of Message Handling System (MHS) to send his/her text messages to user via telephone line using synthetic voice. In the X.400 MHS recommendation, the protocols and service elements are not specified for the voice message delivery system. Thus, we defined access protocol and service elements for Voice Access Unit based on the application program interface for message transfers between X.400 Message Transfer Agent and Voice Access Unit. The system architecture and operations will be provided.

In this paper, the influence of voice source estimation and modeling on speech synthesis and coding is examined and then their new estimation and modeling techniques are proposed and verified by computer simulation. It is known that the existing speech synthesizer produced the speech which is dull and inanimated. These problems are arised from the fact that existing estimation and modeling techniques can not give more accurate voice parameters. Therefore, in this paper we propose a new voice source estimation algorithm and modeling techniques which can not give more accurate voice parameters. Therefore, in this paper we propose a new voice source estimation algorithm and modeling techniques which can represent a variety of source characteristics. First, we divide speech samples in one pitch region into four parts having different characteristics. Second, the vocal-tract parameters and voice source waveforms are estimated in each regions differently using sequential SVD. Third, we propose composite source model as a new voice source model which is represented by weighted sum of pre-defined basis functions. And finally, the weights and time-shift parameters of the proposed composite source model are estimeted uning EM(estimate maximize) algorithm. Experimental results indicate that the proposed estimation and modeling methods can estimate more accurate voice source waveforms and represent various source characteristics.

This paper describes two methods to assess the output speech, quality of vocoders for telephone conferencing in digital mobile communication networks. The proposed methods are the sentence discrimiantion method and the modified degraded mean opinion score (MDMOS) test. We apply these two methods to Qualcomm code excited linear prediction (QCELP), vector sum excited linear prediction (VSELP) and regular pulse excited-long term predictin (RPE-LTD) voceders to evaluate which vocoding algorithm can process mixed voice signal from two speakers better for telephone conferencing. From the experiments we obtain that the VSELP vocoding algorithm reveals superior output speech quality to the other two.

Code Excited Linear Prediction (CELP) speech coders exhibit good performance at data rates below 4.8 kbps. The major drawback to CELP type coders is their many computation. In this paper, we propose a new pitch search method that preserves the quality of the CELP vocoder with reducing complexity. The basic idea is to apply the preprocessing technique beforehand grasping the autocorrelation property of speech waveform. By using the proposed method, we can get approximately 77% complexity reduction in the pitch search.

Poor speech intelligibility in an air traffic control room is frequently a result of many, quite different causes and occasionally leads to complaints of the controller personnel. The paper describes a sequence of successful tasks performed in a local control room. The initial measurements included an investigation of the background noise (caused by fans, air condition, computer and radar equipment) and performance checks of the electronic audio and communication equipment with respect to the audio transmission behavior. The spectral composition of the noise as well as the characteristics of the audio communication path between the controllers and the pilots(which showed a loss of spectral information in the audio band due to built-in notch filters for the suppression of control tones) required adaptations of the amplitude behavior of the amplifiers through user adjustable tone controls. The radar console fans, which contributed significantly to the overall noise floor of the room, underwent a substantial reconstruction by replacing the tight mounting with an elastic double suspension, reducing the noise level by 50%. Finally, a possible source of untimely fatigue of the controllers during their working hours has been found in strong spectral components of the noise above the audio band, radiated by numerous video monitors in the control through vibrating components excited by the line frequency of the video signal.

The use of adaptive feedback cancellation to prevent howling requires a reference signal that is correlated with the feedback signal by is not correlated with the input signal. Such a signal is hard to obtain in hearing aids. In this paper, the use fo frequency compression to decorrelate the output signal with input signal for use as reference is presented. Performance evaluation results indicate that with the proper choice of system parameters, the use of this system can provide a significant increase in howling margin with minimal deterioration in output signal quality.

We consider statistical properties of the generalized Capon's method. It is observed that the estimation error of the generalized Capon's method has almost the same variance as the MUSIC method, although the generalized Capon's method yields a slightly biased estimate.

Although an ideal low-pass filter is not physically realizable, it can be approximated on the basis of time reversal techniques. In this paper, we describe a method to approximately implement the ideal low-pass filter and apply it to the pitch extraction system. Experimental results show that our method is effective to estimate the fundamental frequency of the speech signal.

Text-to-Speech(TTS) conversion system can convert any words or sentences into speech. To synthesize the speech like human beings do, careful prosody control including intonation, duration, accent, and pause is required. It helps listeners to understand the speech clearly and makes the speech sound more natural. In this paper, a prosody control scheme which makes use of the information of the function word is proposed. Among many factors of prosody, intonation, duration, and pause are closely related to syntactic structure, and their relations have been formalized and embodied in TTS. To evaluate the synthesized speech with the proposed prosody control, one of the subjective evaluation methods-MOS(Mean Opinion Score) method has been used. Synthesized speech has been tested on 10 listeners and each listener scored the speech between 1 and 5. Through the evaluation experiments, it is observed that the proposed prosody control helps TTS system synthesize the more natural speech.

The vector sum excited linear prediction(VSELP) coding gives high quality of synthetic speech at bit rates as low as 4.8kbps, but its computational complexity is prohibitive for real time applications. In this paper, we propose a method to reduce the computations of the VSELP codebook search procedure. The proposed method reduces the search space efficiently, before applying every linear combination of the basis vectors to the codebook search procedure. It decides whether is can fix the combination coefficient of each basis vector using heuristics so that the number of combinations decreases. It has been shown that the proposed method retains good quality of synthetic speech and reduces the computations of codebook search procedure by more than 40% of the origin.

In this paper, we introduce a new speech synthesis method for Japanese and Korean arbitrary sentences using the natural speech data-base. Also, application of this method to a CAI system is discussed. In our synthesis method, a basic sentence and basic accent-phrases are selected from the data-base against a target sentence. Factors for those selections are phrase dependency structure (separation degree), number of morae, type of accent and phonemic labels. The target pitch pattern and phonemic parameter series are generated using those selected basic units. As the pitch pattern is generated using patterns which are directly extracted form real speech, it is expected to be more natural than any other pattern which is estimated by any model. Until now, we have examined this method on Japanese sentence speech and affirmed that the synthetic sound preserves human-like features fairly well. Now we extend this method to Korean sentence speech synthesis. Further more, we are trying to apply this synthesis unit to a CAI system.

Real-time RELP vocoder is implemented on the TMS320C25 DSP chip. The implemented system is IBM-PC add-on board and composed of analog in/out unit, DSP unit, memoy unit, IBM-PC interface unit and its supporting assembly software. Speech analyzer and synthesizer is implimented by DSP assembly software. Speech parameters such as LPC coefficients, base-band residuals, and signal gains is extracted by autocorrelation method and inverse filter and synthesized by spectral folding method and direct form synthesis filter in this board. And then, real-time RELP vocoder with 9.6Kbps is simulated by down-loading method in the DSP program RAM.

In speech signal processing, it is very important to detect the pitch exactly. The algorithms for pitch extraction that have been proposed until now are not enough to detect the fine pitch in speech signal. Thus we propose the new algorithm which takes advantage of the G-peak extraction. It is the method to find MZCI(maximum zer-crossing interval) which is defined as cut-off bandwidth rate of LPF (low pass filter)and detect the pitch period of the voiced signals. This algorithm performs robustly with a gross error rate of 3.63% even in 0 dB SNR environment. The gross error rate for clean speech is only 0.18%. Also it is able to process all course with speed.

A speech display system is developed for the evaluation and the training of speech utterance. The speech is analyzed by linear predictive technique every 5 ms and the frequencies of the lowest two spectral local peaks P1 and P2 are extracted. The vowel trakectory is displayed using those frequencies on th P1-P2 plane. In most cases, P1 and P2 correspond to the first and the second formants, but in the case of indistinct utterance, the correspondence between the local spectral peaks and the formants tends to fall into disorder. And the system is considered to be useful for the evaluation of speech quality. The examples of some words uttered by normal speakers and some patients with difficulty in utterance are compared each other for the discussion of the effectiveness of the system.

In this paper we have evaluated the synthetic speech quality by the proposed TD-PCULI speech synthesis method. For the synthesis we have extracted parameters from the Korean monosyllables through the analysis of speech waveforms in the time domain. We have constructed the Korean data format dictionary for the synthesis-by-rule depending upon the frequencies of the Korean pronunciation large vocabulary dictionary, in which V type syllables are 19, CV type's are 80, VC type's are 30 and CVC type's are 100. And using them we have synthesized various Korean monosyllables, words and sentences. We have tested each 10 syllables selected according to the 4 Korean syllable types with the objective MOS(Mean Opinion Score) evluation method about the 4 items i.e., intelligibility, clearness, loudness, and naturality after selecting random group without the knowledge of them. And also we have tested the possibility to modify a duration and F0 into another forms with changing a duration (i.e., 150msec, 300msec, 500msec, 700msec and 1sec) and a central fundamental frequency(i.e., 80Hz, 118Hz, 140Hz, 170Hz, and 200Hz). As the results of experiments the noises occurred in the course of synthesizing the speech by the rules are removed to be a very clear level and we can find that the prosodic elements can be controled as a good condition.

In this paper a new speech synthesis method in the time domain using mono-syllables is proposed. It is to overcome the degradation of the synthetic speech quality by the synthesis method in the frequency domain and to develop an algorithm in the time domain for the prosodic control. In particular when we use a method in a time domain with mono-syllable as a synthesis unit it will be the main issues which are to control th pitch period and to smooth the energy pattern. As a solution to the pitch control, a method using Lagrange interpolation is suggested. As a solution to the other problem, an algorithm which can control the amplitude envelop shape of mono-syllable is proposed. As the results of experiments it was possible to synthesize unlimited Korean speeches including the prosody control. Accoding to the MOS evaluation the quality and the naturality in them was improved to be a good level.

Surrounding noise often affects the performance of speech recognition system when it is used in office or home. Especially situation is more serious when colored and nonstational noise such as an sound from television or other audio equipment is introduced. The authors proposed a voice control system for television set using an adaptive noise canceler, and it works well even is sound of television set has comparable level of speech. In this paper, a new front-end of speech recognition is introduced for the voice control system. This font-end utilizes a simplified masking model to reduce the effect of residual noise. According to experimental results, 90% correct recognition is achieved even if the level of television sound is almost 15dB higher than one of speech.

Predictive neural network models are powerful speech recognition models based on a nonlinear pattern prediction. Those models can effectively normalize the temporal and spatial variability of speech signals. But those models suffer from poor discrimination between acoustically similar words. In this paper, we propose a discriminative training algorithm for predictive neural network models based on a generalized probabilistic descent (GPD) algorithm and minimum classification error formulation (MCEF). The Evaluation of our training algorithm on ten Korean digits shows its effectiveness by 40% reduction of recognition error.

This paper is about an experiment of speaker-independent automation Korean spoken words recognition using Multi-Layered Perceptron and Error Back-propagation algorithm. The words were not segmented into syllables or phonemes, and some feature components extracted from the words in equal gap were applied to the neural network. This paper tried to find out the optimum conditions through various experiment which are comparison between total and pre-classified training.

This paper improves the previously proposed spectral mapping method for supervised speaker adaptation in which a mapped spectrum is interpolated from speaker difference vectors at typical spectra based on a minimized distortion criterion. In estimating these difference vectors, it is important to find an appropriate number of typical points. The previous method empirically adjusts the number of typical points, while the present method optimizes the effective number by rank reduction of normal equation. This algorithm was applied to a supervised speaker adaptation for Korean word recognition using the templates form a prototype male speaker. The result showed that the rank reduction technique not only can automatically determine an optimal number of code vectors, but also slightly improves the recognition scores compared with those obtained by the previous method.

This paper presents the Korean digit recognition method under noise environment using the spectral mapping training based on static supervised adaptation algorithm. In the presented recognition method, as a result of spectral mapping from one space of noisy speech spectrum to another space of speech spectrum without noise, spectral distortion of noisy speech is improved, and the recognition rate is higher than that of the conventional method using VQ and DTW without noise processing, and even when SNR level is 0 dB, the recognition rate is 10 times of that using the conventional method. It has been confirmed that the spectral mapping training has an ability to improve the recognition performance for speech in noise environment.

This paper proposes a filter structure suitable for speech synthesis applications. We first derive the lossy pole-zero model by employing the wave digital filter(WDF) adaptor formula, and by converting the fixed termination value - 1 into a loss factor $\mu$c$\in$(-1, 1). Then we discuss how to determine the reflection We employ the Durbin's method in estimating the numerator polynomial of the lossy pole-zero transfer function from the given speech sound, and then apply the step-down algorithm on the numerator to extract the reflection coefficients of the closed-termination tract. For determining the reflection coefficients of the other parts we employ a pre-calculated pole-estimator polynomial.

This paper introduces a interword modeling and a Viterbi search method for continuous speech recognition. We also describe a development of a real-time voice dialing system which can recognize around one hundred words and continuous digits in speaker independent mode. For continuous digit recognition, between-word units have been proposed to provide a more precise representation of word junctures. The best path in HMM is found by the Viterbi search algorithm, from which digit sequences are recognized. The simulation results show that a interword modeling using the context-dependent between-word units provide better recognition rates than a pause modeling using the context-independent pause unit. The voice dialing system is implemented on a DSP board with a telephone interface plugged in an IBM PC AT/486.

This paper describes recognition results using the modified Learning Vector Quantization (MLVQ2) method which we proposed previously. At first, we investigated the property of duration of 29 Korean consonants and found that the variances of th duration were extremely big comparing to other languages. We carried out preliminary recognition experiments for three stop consonants P, T and K. From the recognition results, we defined the optimum conditions for the learning. Then we applied the MLVQ2 method to the recognition of Korean consonants. The training was carried out using the phoneme samples in the 611 word vocabulary uttered by 2 male speakers, where each of the speakers uttered two repetitions. The recognition experiment was carried out for the phoneme samples in two repetitions of the 611 word vocabulary uttered by another male speaker. The recognition scores for the twelve plosives were 68.2% for the test samples. The recofnition scores for the 29 Korean consonants were 64.8% for the test samples.

The speech recognition systems using VQ have usually the problem decreasing recognition rate, MSVQ assigning the dissimilar vectors to a segment. In this paper, applying One-stage DMS/DP algorithm to the recognition experiments, we can solve these problems to what degree. Recognition experiment is peformed for Korean DDD area names with DMS model of 20 sections and word unit template. We carried out the experiment in speaker dependent and speaker independent, and get a recognition rates of 97.7% and 81.7% respectively.

It is known that SOFM has the property of effectively creating topographically the organized map of various features on input signals, SOFM can effectively be applied to the recognition of Korean phonemes. However, is isn't guaranteed that the network is sufficiently learned in SOFM algorithm. In order to solve this problem, we propose the learning algorithm combined with the conventional K-means clustering algorithm in fine-tuning stage. To evaluate the proposed algorithm, we performed speaker dependent recognition experiment using six phoneme classes. Comparing the performances of the Kohonen's algorithm with a proposed algorithm, we prove that the proposed algorithm is better than the conventional SOFM algorithm.

Dynamically localized self-organizing map model (DLSMM) is a new speech recognition model based on the well-known self-organizing map algorithm and dynamic programming technique. The DLSMM can efficiently normalize the temporal and spatial characteristics of speech signal at the same time. Especially, the proposed can use contextual information of speech. As experimental results on ten Korean digits recognition task, the DLSMM with contextual information has shown higher recognition rate than predictive neural network models.

In this study we developed monosyllable lists for articulation test for Korean. We sampled 103,581 colloquial monosyllables, applied them to five selection rules that based on Korean linguistic characteristics, and finally constructed five different lists with fifty monosyllables. The validity test using the monaural impairment factors such as S/N ratio and cut-off frequency showed that articulation scores were chanted systematically according to the level of impairment factors. In addition, we investigated the effect of azimuth of a single competing sound upon articulation scores. The syllables were always reproduced by the loudspeaker in front of the subject, while Hoth noise were reproduced by the loudspeaker with varying azimuth around subject. The result indicated that the articulation depended on the azimuth of competing sound sources. Finally, no significant differences among lists were found in all experimental conditions.

In order to realize the function of human interface of telecommunications whose objective is to interchange useful information among persons, we developed a bone conduction telephone with which hearing impaired persons with conductive or noise-induced hearing loss and presbycusis can communicate with each other without any other additional devices such as hearing aids. The bone conduction telephone we developed has chatacteristics as follows : (i) a hearing impaired person and a normal hearing person can communicate by bone and air conduction hearings, respectively, using only this telephone set because, as its receiver, it uses a bone conduction vibrator with which we can realize such function with the voice coil and damper of a small speaker unit, the vibrating plate, etc., (ii) it has tone control function compensating hearing losses of hearing impaired persons according to their hearing loss/frequency chatacteristics. Using the tone control function together with a received volume control, it has the received volume range of 20dB in loudness rating; and (iii) it has the function of three emergency calls and a bell lamp as the visual display of a received call.

This paper describes the method for designing loudness ratings as transmission quality for ISDN telephone connected to fully digital network. To design the desirable loudness ratings for ISDN telephone, the model system of digital speech communication for subjective test is developed and opinion tests for establishing the optimal CODEC input level, the range of overall loudness rating, and sidetone masking rating are performed. As the results, the desirable ranges of loudness ratings are proposed as 6 to 8dB for sending, 0 to 2dB for receiving, and 10 to 14dB for sidetone masking rating.

This paper presents the linguistic processing for the Automatic Interpretation system between English/Korean language pair. We introduce two machine translation systems, each for English-to-Korean and Korean-to-English, describe the system configuration and several characteristics, and discuss the translation evaluation results.

A set of text database is indispensable to the probabilistic models for speech recognition, linguistic model, and machine translation. We introduce an environment to canstruct text databases : an automatic tagging system and a set of tools for lexical knowledge acquisition, which provides the facilities of automatic part of speech recognition and guessing.

This paper presents a method to predict in statistics the coded signal propagation in fading channel with the help of the ray theory. This method features its high speed and efficiency. The predictions of received signal envelope and pulse width can be give out quickly.

To solve the frequency variation of speech patterns which consist of LPC sequences, a new membership function made by the relation between order of LPC and spectrum is proposed in this paper. To reduce errors, fuzzy inference is executed using the proposed membership function. The computer simulation shows the effectiveness of the word recognition.

In this paper, we design and implement the Korean speech synthesis by rule system. This system is applied the multiband excitation signal on voiced sounds. The multiband excitation signal is obtained by mixing impluse spectrum and which noise spectrum. We find that the quality of synthesized speech is improved using this application. Also, we classify the voiced sounds by cepstral euclidian distance measure for reducing overhead memory. The representative excitation signal of the same group's voiced sounds is used as excitation signal on synthesis. This method does not affect the quality of synthesized speech. As the result of experiment, this method eliminates the "buzziness" of synthesized speech and reduces the spectral distortion of synthesized speech.ed speech.

Code Excited Linear Prediction(CELP) as a speech coder exhibits good performance at data rates below 4.8 kbps. The major drawback to CELP type coders is their large amount of computation. In this paper, we propose a new pitch search method that preserves the quality of the CELP vocoder with reduced complexity. The basic idea is to restrict the pitch searching range by estimating the preliminary pitches. Applying the proposed method to the CELP vocoder, we can get approximately 87% complexity reduction in the pitch search.