A new method for the compression of electrocardiographic (ECG) data is presented. The method is based on the orthonormal wavelet analysis recently developed in applied mathematics. By using wavelet transform, the original signal is decomposed into a set of sub-signals with different frequency channels corresponding to the different physical features of the signal. By utilizing the optimum bit allocation scheme, each decomposed sub-signal is treated according to its contribution to the total reconstruction distortion and to the bit rate. In our experiments, compression ratios (CR) from 13.5: 1 to 22.9: 1 with the corresponding percent rms difference (PRD) between 5.5% and 13.3% have been obtained at a clinically acceptable signal quality. Experimental results show that the proposed method seems suitable for the compression of ECG data in the sense of high compression ratio and high speed.

A module for rule-based Japanese speech synthesis has been developed. The synthesizer was constructed using the Multiple-Cascade Terminal Analog (MCTA) structure, and this sturcture has been improved in three respects: the voicing-source model has an increased number of variable parameters which allows for voicing-source waveforms that better approximate natural speech; the spectral characteristics of the fricative source have been improved; and the path used for nasal consonants has an increased number of resonators to better conform to theory. The current synthesis system uses a modified stored-pattern data structure which allows better transitions between syllables; however, time-invariant values are used in certain cases in order to decrease the amount of required memory. This system also has a new consolidated method for generating geminate obstruents and syllabic nasals. This synthesizer and synthesis system have been implemented in a re-developed rule-based speech-synthesis module. This module has been constructed using ASIC technology and has both small size (56368 mm) and light weight (19g); it is therefore possible to embed it in various types of portable or moving machinery. The module can be connected directly to a mocroprocessor bus and accepts as input sentences which are generated by the host computer. The input sentences are written with the Japanese katakana or romaji syllabaries and other symbols which describe the sentence structure. The syllable articulation rate for one hundred Japanese syllables (including palatalized sounds) is 65% and for sixty-seven syllables (not including palatalized sounds) is 74%. The word intelligibility, measured using phonetically-balanced words, it 88%.

A terminal-analog synthesizer which uses a glottal model has already been proposed for rule-based speech synthesis, but the control strategy for glottal source intensity levels has not yet been defined. On the other hand, power-control rules which determine the target segmental power of synthetic speech have been proposed, based on statistical analysis of the power in natural speech. It is pointed out that there is a close correlation between observed fundamental frequency and power levels in natural speech; however, the theoretical reasons for this correlation have not been explained. This paper shows the relationship between fundamental frequency and resultant power in a terminal-analog synthesizer which uses a glottal model. From the equations it can be deduced that the tendency in natural speech for power to increase with fundamental frequency can be closely simulated by the sum of the effect of the radiation characteristic and the effect of the synthesis system's vocal tract transfer function. In addition, this paper proposes a method for adjusting the power of synthetic speech to any desired value. This control method can be executed in real-time.

We deal with a new type ceramic emitter which is used in a cleanroom ionizer system and is composed of a needle-shaped silicon and a rod-shaped silicon carbide ceramics. The discharge test was carried out to investigate the particle generation from the emitter and the degradation of the emitter. As a result, it was found that the ceramic emitter had practically higher performance than a conventional tungsten emitter.

The authors developed a portable Japanese text-to-speech system using a pocket-sized formant speech synthesizer. It consists of a linguistic processor and an acoustic processor. The linguistic processor runs on an MS-DOS personal computer and has functions to determine readings and prosodic information for input sentences written in kana-kanji-mixed style. New techniques, such as minimization of a cost function for phrases, rare-compound flag, semantic information, information of reading selection and restriction by associated particles, are used to increase the accuracy of readings and accent positions. The accuracy of determining readings and accent positions is 98.6% for sentences in newspaper articles. It is possible to use the linguistic processor through an interface library which has also been developed by the authors. Consequently, it has become possible not only to convert whole texts stored in text files but also to convert parts of sentences sent by the interface library sequentially, and the readings and prosodic information are optimized for the whole sentence at one time. The acoustic processor is custom-made hardware, and it has adopted new techniques, for the improvement of rules for vowel devoicing, control of phoneme durations, control of the phrase components of voice fundamental frequency and the construction of the acoustic parameter database. Due to the above-mentioned modifications, the naturalness of synthetic speech generated by a Klatt-type formant speech synthesizer was improved. On a naturalness test it was rated 3.61 on a scale of 5 points from 0 to 4.

A text-to-speech conversion system for Japanese has been developed for the purpose of producing high-quality speech output. This system consists of four processing stages: 1) linguistic processing, 2) phonological processing, 3) control parameter generation, and 4) speech waveform generation. Although the processing at the first stage is restricted to the texts on general weather conditions, the other three stages can also cope with texts of news and narrations on other topics. Since the prosodic features of speech are largely related to the linguistic information, such as word accent, syntactic structure and discourse structure, linguistic processing of a wider range than ever, at least a sentence, is indispensable to obtain good quality speech with respect to the prosody. From this point of view, input text was restricted to the weather forecast sentences and a method for linguistic processing was developed to conduct morpheme, syntactic and semantic analyses simultaneously. A quantitative model for generating fundamental frequency contours was adopted to make a good reflection of the linguistic information on the prosody of synthetic speech. A set of prosodic rules was constructed to generate prosodic symbols representing prosodic structures of the text from the linguistic information obtained at the first stage. A new speech synthesizer based on the terminal analog method was also developed to improve the segmental quality of synthetic speech. It consists of four paths of cascade connection of pole/zero filters and three waveform generators. The four paths are respectively used for the synthesis of vowels and vowel-like sounds, nasal murmur and buzz bar, friction, and plosion, while the three generators produce voicing source waveform approximated by polynomials, white Gaussian noise source for fricatives and impulse source for plosives. The validity of the approach above has been confirmed by the listening tests using speech synthesized by the developed system. Improvements both in the quality of prosodic features and in the quality of segmental features were realized for the synthetic speech.

We present an extension of the previously proposed 3-layer feedforward network called a cascaded network. Cascaded networks are trained to realize category classification employing binary input vectors and locally represented binary target output vectors. To realize a nonlinearly separable task the extended cascaded network presented here is consreucted by introducing high order cross producted inputs at the input layer. In the construction of the cascaded network, two 2-layer networks are first trained independently by delta rule and then cascaded. After cascading, the intermediate layer can be understood as a hidden layer which is trained to attain preassigned saturated outputs in response to the training set. In a cascaded network trained to categorize binary image patterns, saturation of hidden outputs reduces the effect of corrupted disturbances presented in the input. We demonstrated that the extended cascaded network was able to realize a nonlinearly separable task and yielded better generalization ability than the Backpropagation network.

Evolution programs have been shown to be very useful in a variety of search and optimization problems, however, until now, there has been little attempt to apply evolution programs to channel routing problem. In this paper, we present an exolution program and identify the key points which are essential to successfully applying evolution programs to channel routing problem. We also indicate how integrating heuristic information related to the problem under consideration helps in convergence on final solutions and illustrate the validity of out approach by providing experimental results obtained for the benchmark tests. compared with the optimal solutions.

The performance of ASIPs (Application Specific Integrated Processors) is heavily affected by the design of their instruction set architecture. In order to maximize the performance of ASIP, it is essential to design an architecture that has an optimum instruction set. This paper descibes a new method that automates the design of optimum instruction set of ASIP. This method solves the Instruction set implementation Method Selection Problem(IMSP). IMSP is to be solved in the instruction set architecture design. Frse, the IMSP is formalized as an integer programming problem, which is to maximize the perfomance of the CPU under the constraints of chip area and power consumption. Then, a branch-and-bound algorithm to solve IMSP is described. According to the experimental results, the proposed algorithm is quite effective and efficient in solving the IMSP. The presented method automates a complex part of the ASIP chip design and is also a good design tool that enables designer to predict the performance of their design before completion.

One of the most interesting and most analyzed aspects of the CPU design is the instruction set design. How many and which operations to be provided by hardware is one of the most fundamental issues relaing to the instruction set design. This paper describes a novel method that formulates the instruction set design of ASIP (an Application Specific Integrated Processor) using a combinatorial appoach. Starting with the whole set of all possible candidata instructions that represesnt a given application domain, this approach selects a subset that maximizes the performance under the constraints of chip area, power consumption, and functional module sharing relation among operations. This leads to the efficient implementation of the selected instructions. A branch-and-bound algorithm is used to solve this combinatorial optimization problem. This approach selects the most important instructions for a given application as well as optimizing the hardware resources that implement the selected instructions. This approach also enables designers to predict the perfomance of their design before implementing them, which is a quite important feature for producing a quality design in reasonable time.

As in other fields, the automatization of railway maintenance work is a firm requirement. The authors have developed a system detecting obstacles around a railway for practical railway inspection. The system is based on an original laser-sectioning method and characterized by high accuracy with wide view and in-motion operation. It was confirmed that a static calibration was performed at an accuracy of within 5 mm. Furthermore, a theoretical estimation predicted that dynamic errors can be eliminated within a resolution of 4 mm by means of rail movement detection. In field tests on the Chuo Line, facilities were successfully inspected at speeds up to 40km/h.

Scattering characteristics of a stratified chiral slab, which is composed of dielectric chiral layers of different material properties and thicknesses, are extensively explored. Design considerations for optical filters are also presented for both the cases of normal and oblique incidence. In the analysis, the incident field is assumed to be a plane monochromatic wave of any arbitrary polarization. The transmitted and reflected electric fields are obtained by noting the fact that the electric field inside a chiral medium is expressed as a sum of the left- and right-circularly polarized plane waves of different phase velocities. Then one can describe the power densities and the Stokes parameters of the transmitted and reflected waves in terms of their field components. As is well known,the Stokes parameters characterize every possible state of polarization of a plane wave. Numerical examples are presented to show the effects of chirality on polarization conversion properties of the stratified chiral slab. The cross- and co-polarized powers and the Stokes parameters of the transmitted and reflected waves are computed for the incident wave of perpendicular polarization. The numerical results demonstrate novel depolarization properties of the slab with application to the design of efficient filters active at the optical region. It is seen from the results that the stratified chiral slab acts as a polarization-conversion transmission filter that passes only a cross-polarized component of the transmitted wave at some frequency band. Furthermore, the slab may be utilized as an antireflection filter for both the cross- and co-po1arized components of the reflected wave at the band region.

This letter presents a new method for adaptive control of nonminimum phase discrete-time systems with disturbances based on the technique of pole-zero placement. The long division method is used to decompose apolynomial into a stable polynomial and an unstable one. Finally, the results of computer simulation are presented to illustrate the effectiveness of the proposed method.

A coherent communication system using squeezed light is one of candidates for a realization of super-reliable systems. In order to design such a system, it is essential to understand and to analyze modulators mathematically. However, quantum noise of squeezed light has a colored spectrum which changes with respect to phase of a local laser. Therefore the optimization of the relationship between signal and quantum noise spectrums is required at a modulator to obtain the ultimate performance of the communication system. In this paper, some ideas of modulators for squeezed light are proposed and their spectrum transformations are given. After the brief summary of squeezed quantum noise, a new concept which originates from the restriction of the local laser phase is applied to it. This concept makes a problem originated from a colored quantum noise spectrum more serious. It results in the optimization problem for the relationship between the quantum noise spectrum and signal power spectrum. The solution of this problem is also given under the restriction of local laser phase. As a result, a general design theory for coherent communication system using the squeezed light is given.

This paper tends to analyze the trends of the research in logic synthesis. The first part is devoted to an expertise of the efficiency of factorization methods developed during the last decade and to the proposal of dedicated methods for complex logic blocks. The second part shows the importance of Binary Decision Diagrams as representation of Boolean functions. Their use in the technology mapping phase of multiplexor-based FPGAs in an industrial tool is taken as illustration.

This paper first proposes a fast fingerprint identification method based on a weighted-mean of binary image, and further investigates optimization of the weights. The proposed method uses less computer memory than the conventional pattern matching method, and takes less computation time than both the feature extraction method and the pattern matching method. It is particularly effective on the fingerprints with a small angle of inclination. In order to improve the identification precision of the proposed basic method, three schemes of modifying the proposed basic method are also proposed. The performance of the proposed basic method and its modified schemes is evaluated by theoretical analysis and computer experiment using the fingerprint images recorded from a fingerprint read-in device. The numerical results showed that the proposed method using the modified schemes can improve both the true acceptance rate and the false rejection rate with less memory and complexity in comparison with the conventional pattern matching method and the feature extraction method.

This paper presents a delivery mechanism using a spectrum delivery switch (SDS) in a microcell system. In our fiber-optic microcell systems, modulators, demodulators and spectrum delivery switches are installed in a central station. A spectrum delivery switch controls provide flexible dynamic channel assignment and functions as a hand over algorithm. This control method employs a TDMA time slot switch and a MODEM connection switch. The relation between blocking probability and offered traffic are described and computer simulation results are shown. The results indicate an improvement in this blocking probability over conventional systems.

This paper presents two new maximum likelihood decoding (MLD) algorithms for linear codes over Z-channel, which are much more efficient than conventional exhaustive algorithms for high rate codes. In the proposed algorithms, their complexities are reduced by employing the projecting set Cs of the code, which is determined by the "projecting" structure of the code. Space and computational complexities of algorithms mainly depend upon the size of Cs which is usually several times smaller than the total number of codewords. It is shown that the upper bounds on computational complexities of decoding algorithms are in proportion to the number of parity bits and the distance between an initial estimate of the codeword and the received word, respectively, while space complexities of them are equal to the size of Cs. Lastly, numerical examples clarify the average computational complexities of the proposed algorithms, and the efficiency of these algorithms for high rate codes is confirmed.

Asymptotic bounds are considered for unidirectional byte error-correcting codes. Upper bounds are developed from the concepts of the Singleton, Plotkin, and Hamming bounds. Lower bounds are also derived from a combination of the generalized concatenated code construction and the Varshamov-Gilbert bound. As the result, we find that there exist codes of low rate better than those on the basis of Hamming distance with respect to unidirectional byte error-correction.

Amorphous films of YBCO and BSCCO (2212) sputtered on MgO substrate were crystallized using zonemelt technique. For YBCO films, thin Ag intermediate layer was found to be effective in enhancing crystal growth and preferred orientation. Zone-melted BSCCO films included both (2201) phase and Cu(Sr, Ca)O2 in a form of dendritic crystallites. Tc's obtained for YBCO and BSCCO films were 70 and 75 K, respectively.