This paper proposes a reconfigurable parallel processor based on a two-dimensional linear celular automaton model. The processor based on the model can be reconfigured quickly by utilizing the characteristics of the automaton used for its model. Moreover, the processor has short data path length between processing elements compared with the length of the processor based on one-dimensional linear cellular automaton model which has been already discussed. The processing elements of the processor based on the two-dimensional linear cellular automaton model are regarded as cells and the operational states of the processor are treated as the states of the automaton. When faults are detected, the processor can be reconfigured by changing its state under the state transition function of the processor determined by the weighting function of the automaton model. The processor can be reconfigured within a clock period required for making a state transition. This processor is extremely effective for real-time data processing systems required high reliability.

For the production of high quality synthetic sounds in a text-to-speech system, an excellent synthesizing method of speech signals is indispensable. In this paper, a new speech analysis-synthesis method for the text-to-speech system is proposed. The signals of voiced speech, which have a line spectrum structure at intervals of pitch in the linear frequency domain, can be represented approximately by the superposition of sinusoidal waves. In our system, analysis and synthesis are performed using such a harmonic structure of the signals of voiced speech. In the analysis phase, assuming an exact harmonic structure model at intervals of pitch against the fine structure of the short-time power spectrum, the fundamental frequency f0 is decided so as to minimize the error of the log-power spectrum at each peak position. At the same time, according to the value of the above minimized error, the rate of periodicity of the speech signal is detemined. Then the log-power spectrum envelope is represented by the cosine-series interpolating the data which are sampled at every pitch period. In the synthesis phase, numerical solutions of non-linear differential equations which generate sinusoidal waves are used. For voiced sounds, those equations behave as a group of mutually synchronized oscillators. These sinusoidal waves are superposed so as to reconstruct the line spectrum structure. For voiceless sounds, those non-linear differential equations work as passive filters with input noise sources. Our system has some characteristics as follows. (1) Voiced and voiceless sounds can be treated in a same framowork. (2) Since the phase and the power information of each sinusoidal wave can be easily controlled, if necessary, periodic waveforms in the voiced sounds can be precisely reproduced in the time domain. (3) The fundamental frequency f0 and phoneme duration can be easily changed without much degradation of original sound quality.

Material research on capacitor dielectrics for DRAM applications is reviewed. The state of the art technologies to prepare Si3N4,Ta2O5, and SrTiO3 thin films for capacitors are described. The down-scaling limits for Si3N4 and Ta2O5 capacitors seem to be 3.5 and 1.5 nm SiO2 equivalent thickness, respectively. Combined with a rugged polysilicon electrode surface,Si3N4 and Ta2O5 based-capacitors are available for 256 Mbit and 1 Gbit DRAMs. At the present time, the minimum SiO2 equivalent thickness for high permittivity materials is around 1 nm with the leakage current density of 10-7 A/cm2. Among the great variety of ferroelectrics, two families of materials,i.e., Pb (Zr, Ti) O3 and (Ba, Sr) TiO3 have emerged as the most promising candidates for 1 Gbit DRAMs and beyond. If the chemical vapor deposition technology can be established for these materials, capacitor dielectrics should not be a limiting issue for Gbit DRAMs.

In this paper, we quantitively analyzed speech data in seven different styles to make natural Japanese conversational speech synthesis. Three reading styles were produced at different speeds (slow, normal and fast), and four speaking styles were produced by enacting conversation in different situations (free, hurried, angry and polite). To clarify the differences in prosodic characteristics between conversational speech and read speech, means and standard deviations of vowel duration, vowel amplitude and fundamental frequency (F0) were analyzed. We found large variation in these prosodic parameters. To look more precisely at the segmental duration and segmental amplitude differences between conversational speech and read speech, control rules of prosodic parameters in reading styles were applied to conversational speech. F0 contours of different speaking styles are superposed by normalizing the segmental duration. The differences between estimated values and actual values were analyzed. Large differences were found at sentence final and key (focused) phrases. Sentence final positions showed lengthening of segmental vowel duration and increased segmental vowel amplitude. Key phrase positions featured raising F0.

Prosodic features of the spoken Japanese play an important role in the transmission of linguistic information concerning the lexical word accent, the sentence structure and the discourse structure. In order to construct prosodic rules for synthesizing high-quality speech, therefore, prosodic features of speech should be quantitatively analyzed with respect to the linguistic information. With a special focus on the fundamental frequency contour, we first define four prosodic units for the spoken Japanese, viz., prosodic word, prosodic phrase, prosodic clause and prosodic sentence, based on a decomposition of the fundamental frequency contour using a functional model for the generation process. Syntactic units are also introduced which have rough correspondence to these prosodic units. The relationships between the linguistic information and the characteristics of the components of the fundamental frequency contour are then described on the basis of results obtained by the analysis of two sets of speech material. Analysis of weathercast and newscast sentences showed that prosodic boundaries given by the manner of continuation/termination of phrase components fall into three categories, and are primarily related to the syntactic boundaries. On the other hand, analysis of noun phrases with various combinations of word accent types, syntactic structures, and focal conditions, indicated that the magnitude and the shape of the accent components, which of course reflect the information concerning the lexical accent types of constituent words, are largely influenced by the focal structure. The results also indicated that there are cases where prosody fails to meet all the requirements presented by word accent, syntax and discourse.

The intensity-dependent transmission characteristics of an Ag+Na+ ion-exchanged glass waveguide with a nematic liquid crystal MBBA cover have been investigated experimentally using an Ar+ laser. It is found that the transmission characteristics of the TE1 mode are strongly influenced by temperature. Optical bistability has been observed at a particular temperature. Such the strong temperature dependence is believed to be brought by an increase in ordinary refractive index of the MBBA cover due to temperature rise.

An ECL-CMOS SRAM technology is proposed which features a combination of ECL word drivers, ECL write circuits and low-voltage CMOS cells. This technology assures both ultra-high-speed and high-density. In the ECL-CMOS SRAM,various kinds of noise generated during the write cycle seriously affect the memory performance, because it has much faster access than conventional SRAMs. To overcome this problem, we propose three noise reduction techniques; a noise reduction clamp circuit, an emitter follower with damping capacitor and a twisted bit line structure with "normally on" equalizer. These techniques allow fast accese and cycle times. To evaluate these techniques, a 64-kb SRAM chip was fabricated using 0.5-µm BiCMOS technology. This SRAM has a short cycle time of 2 ns and a very fast access time of 1.5 ns. Evaluation proves the usefulness of these techniques.

This paper presents a high-density multiple-valued content-addressable memory (MVCAM) based on a floating-gate MOS device. In the proposed CAM, a basic operation performed in each cell is a threshold function that is a kind of inverter whose threshold value is programmable. Various multiple-valued operations for data retrieval can be easily performed using threshold functions. Moreover, each cell circuit in the MVCAM can be implemented using only a single floating-gate MOS transistor. As a result, the cell area of the four-valued CAM are reduced to 37% in comparison with that of the conventional dynamic CAM cell.

In this paper the two different paradigms for the design of responsive, i.e., distributed fault-tolerant real-time systems, the event-triggered (ET) approach and the time-triggered (TT) approach, are analyzed and compared. The comparison focuses on the temporal properties and considers the issues of predictability, testability, resource utilization, extensibility, and assumption coverage.

A stored channel simulator for digital mobile radio enviroments is proposed, which enables the field tests in the laboratory under identical conditions, since it can reproduce the actual multipath radio channels by using the channel impulse responses (CIR's) measured in the field. Linear interpolation of CIR is introduced to simplify the structure of the proposed simulator. The performance of the proposed simulator is confirmed by the laboratory tests.

Computing devices have reached data frequencies of 100 MHz, and have created a need for small-amplitude impedance-matched buses. We simulated signal transmission characteristics of two basic driver circuits, push-pull and open-drain,for a synchronous DRAM I/O bus. The push-pull driver caused less signal distortion with parasitic inductance and capacitance of packages, and thus has higher frequency limits than the open-drain GTL type. We describe a bus system using push-pull drivers which operates at over 125 MHz. The bus line is 70 cm with 8 I/O loads distributed along the line, each having 25 nH7pF parasitic inductance and capacitance.

This letter reports in detail on the temperature-dependent signal gain characteristics of Er3+-doped optical fiber amplifiers at signal wavelengths of 1.536µm and 1.552µm. The amplifiers were pumped at 0.825µm in a temperature range of 40 to 200. The signal gain for optimum length at both wavelengths stops increasing and begins to decrease at about 80. In the temperature region below 80, both signal gains increase with fiber temperature for fibers of optimum length or less. A temperature independent length aroud the optimum length is observed from 80 to 200 for both signal wavelengths. Theoretically, the temperature dependence of the signal gain characteristics rerults from the changes in fluorescence, absorption, GSA and ESA cross sections.

The transient scattering of a half sine pulse wave by a conducting rectangular cylinder with an open sidewall is rigorously analyzed by using the point matching method (taking into account the edge condition exactly) combined with the fast inversion of Laplace transform. Numerical results are presented for back scattered and forward scattered responses of the far fields when a half sine pulse is incident on the open side and the closed side of the cylinder. The physical meaning of the transient responses is discussed in detail. The comparison of the responses with those by a perfect conducting rectangular cylinder is presented.

This paper discusses methods and numerical simulations of one and two dimensional profilings for an arbitrary convex conducting target using the electromagnetic backscattering. The inversions for profile reconstructions are based upon the modified extended physical optics method (EPO). The modified EPO method assumes the modified physical optics current properly over the entire surface of conducting scatterers. First, the cross sectional area along a line of sight is reconstructed by performing iteratively the Fourier transform of the backscattering field in the frequency domain. Second, the two dimensional profile is reconstructed by synthesizing the above one dimensional results for several incident angles. Numerical simulation results of the target profiling are shown for spheroids and cone-spheroid.

For each two positive integers r, s, let [1DCM(r)-Time(ns)] ([1NCM(r)-Time(ns)]) and [1DCM(r)-Space(ns)] ([1NCM(r)-Space(ns)]) be the classes of languages accepted in time ns and in space ns, respectively, by one-way deterministic (nondeterministic) r-counter machines. We show that for each X{D, N}, [1XCM(r)-Time(ns)][1XCM(r+1)-Time(ns)] and [1XCM(r)-Space(ns)][1XCM(r+1)-Space(ns)]. We also investigate the relationships between one-way multicounter machines and cooperating systems of one-way finite automata. In particular, it is shown that one-way (one-) counter machines and cooperating systems of two one-way finite automata are equivalent in accepting power.

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.

In this paper, a novel Y-junction type demultiplexer utilizing a stratified-waveguide configuration in the branching region is proposed for the purpose of improving the extinction ratio. A high extinction ratio of about 20 dB is achieved at 0.6328 µm and 0.83 µm operation wavelengths both for the TE and TM modes. The properties of the new type branchig waveguides which consist of the diffused waveguide and the striploaded waveguide are described to explain the operation principle. Simulation results by the BPM are also shown to check the designed values of the waveguide parameters.

We propose a third-order low-pass notch filter realized by a single operational amplifier and a minimum number of equal-valued capacitors. As a design example we realize a Chebyshev filter with a ripple of 0.5 dB and it is shown that the experiment result is very good.

A systematic theory of the optimum sub-band interpolation using parallel wavelet filter banks presented with respect to a family of n-dimensional signals which are not necessarily band-limited. It is assumed that the Fourier spectrums of these signals have weighted L2 norms smaller than a given positive number. In this paper, we establish a theory that the presented optimum interpolation functions satisfy the generalized discrete orthogonality and minimize the wide variety of measures of error simultaneously. In the following discussion, we assume initially that the corresponding approximation formula uses the infinite number of interpolation functions having limited supports and functional forms different from each other. However, it should be noted that the resultant optimum interpolation functions can be realized as the parallel shift of the finite number of space-limited functions. Some remarks to the problem of distinction of images is presented relating to the generalized discrete orthogonality and the reciprocal property for the proposed approximation.

The discrete time analysis of logic circuits is usually more efficient than the continuous time analysis, but the preciseness of the discrete time analysis is not guaranteed. The paper shows a method to decide a unit time for a logic circuit under which the analysis result is the same as the result based on the continuous time. The delay time of an element is specified with an interval between the minimum and maximum delay times, and we assume an analysis method which enumerates all possible delay cases under the deisrete time. Our main theorem is as follows: refine the unit time by a factor of 1/2, and if the analysis result with a unit time u and that with a unit time u/2 are the same, then u is the expected unit time.