A responsive network architecture is essential in future open distributed systems. In this paper, a framework that provides the foundations for a responsive network architecture for an internetworking environment is proposed. It is called the Virtually Separated Link (VSL) model. By incorporating this framework, communication of both data and control information can be completed in bounded time. Consequently, a protocol can initiate a recovery mechanism in bounded time, or allow an application to do the same. Its functionalities augment existing resource reservation protocols that support multimedia communication. An overview of a real-time network protocol that is based on this framework is also presented.

Responsive protocols are communication protocols which ensure timely and reliable recovery when error events occur. Protocol synthesis for design of responsive protocols is to derive a protocol specification based on a service specification. In the previous methods, if the service specification includes simultaneous transmission of primitives from a high layer to a low layer through different service access points, then the derived protocol specification includes protocol errors of unspecified reception caused by message collisions. Also, they only includes a recovery function such as retransmission of messages. This is not enough for recovery from abnormal states due to coordination loss. This paper extends a class of derived protocol specifications to include message collisions which usually occur in real communication protocols. Furthermore, this paper proposes a new method for synthesis of a responsive protocal specification derived from a service specification such that the derived protocol specification is free from protocol erros of unspecified receptions caused by message collisions and includes two recovery functions: message retransmission and checkpoint restart functions.

This paper presents a parallel sorting algorithm which sorts n elements on O(n/w+n log n/p) time using p(n) processors arranged in a 1-dimensional grid with w(n1-ε) buses for every fixed ε>0. Furthermore, it is shown that np elements can be sorted in O(n/w+n log n/p) time on pp (pn) processors arranged in a 2-dimensional grid with w(n1-ε) buses in each column and in each row. These algorithms are optimal because their time complexities are equal to the lower bounds.

New memory-macrocell architecture has been developed to obtain high-performance macrocells with a short design Turn-Around-Time (TAT) in ASIC design. The authors propose row- and column-sliceable macrocell architecture in which only nine kinds of rectangular-functional cells, called leaf-cells, are abutted to form macrocells of any sizes. The row-sliceable structure of peripheral circuits is possible due to a newly-developed channel-embedded address decoder combined with via-hole programming. Macrocell performance, especially access time, is kept at a high level by the distributed driver configuration. Zero address-setup time during write operation is actualized by delaying internal write timing with a new delay circuit. A short design TAT of 30 minutes is accomplished due to the simplicity of both macrocell generation and the checking procedure. The macrocells are designed with gate-array and full-custom style, and fabricated with 0.5 µm CMOS technology.

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.

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.

This paper describes automatic generation of speech synthesis rules which predict a stress level for each bunsetsu in long noun phrases. The rules are inductively inferred from a lot of speech data by using two kinds of tree-based methods, the conventional decision tree and the SBR-tree methods. The rule sets automatically generated by two methods have almost the same performance and decrease the prediction error to about 14 Hz from 23 Hz of the accent component value. The rate of the correct reproduction of the change for adjacent bunsetsu pairs is also used as a measure for evaluating the generated rule sets and they correctly reproduce the change of about 80%. The effectiveness of the rule sets is verified through the listening test. And, with regard to the comprehensiveness of the generated rules, the rules by the SBR-tree methods are very compact and easy to human experts to interpret and matches the former studies.

The semiconductor technology has been progressing with a rate of 4 times per every 3 years, and the semiconductor industry has been expanding with the annual growth rate of around 14% in average. Recently, however, the situation seems to be somewhat changing. This paper investigates the trends in the past of VLSI technologies and performances of VLSI chips, of the R & D and equipment investments, and of the production and design costs. By extrapolating these trends, the future prospects for VLSIs are given in both the technology and the economics. According to these prospects, (1) 1 Gbit DRAMs and 50 M transistor system VLSIs would be realized before 2000, (2) investments for R & D and production equipments will steeply increase up to the unreasonable value, and (3) the delay in demand will become longer, which will make the return on investment difficult. As some of the key issues for overcoming these difficulties, the reduction in the investment and the cost,the alliance, and the market creation are discussed.

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.

A new concept for a positioning satellite system based on a new satellite constellation has been studied. The system needs a minimum of four satellites injected into quasi-geostationary orbit (QGEO) with high inclination. Due to the QGEO characteristic, the satellites are orbiting within continuous visibility range of ground control stations (GCS), from which the satellite time is controlled through the link connections of the feeder and the intersatellite communication (ISC). Consideration is made for the required high accuracy and quality checks against malfunction, wherever the satellites may be positioned. The orbit data processing function, another major function, is performed independently of the time control. The case of global coverage attained by twelve satellites has been studied in this paper. When a constellation of satellites for a global navigation satellite system (GNSS) is designed, conditions to obtain a good geometric dilution of precision (GDOP) at all places and times should be considered. Therefore, the satellites will be spread out in wide directions and are in an asymmetrical arrangement when seen by an observer are considered when setting the parameters of the ephemerides of the constellation. Under the restraints of the designed constellation, the GDOP value distribution for a third of the world map with area time parameters is computed and summarized in histograms for the system evaluation.

Multiplication-accumulation is the basic computation required for image filtering operations. For real-time image filtering, very high throughput computation is essential. This work proposes a hardware algorithm for an application-specific VLSI architecture which realizes an area-efficient high throughput multiplier-accumulator. The proposed algorithm utilizes a priori knowledge of filter mask coefficients and optimizes number of basic hardware components (e.g., full adders, pipeline latches, etc.). This results in the minimum area VLSI architecture under certain input/output constraints.

A quick evaluation method is proposed to obtain stability robustness measures in polynomial coefficient space based on knowledge of coefficients of a Hurwitz stable nominal polynomial. Two norms are employed: l- and l2-norm, which correspond to the stability hypercube and hyperball in the space, respectively. Just inverting Hurwitz matrix for the nominal polynomial immediately yields closed-form estimates for the size of the hypercube and hyperball.

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 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.

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.

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.

To make a fast Bi-CMOS SRAM yield high without speed degradation, three defect-repair methods, the address comparison method, the fuse decoder method and the distributed fuse method, were considered in detail and their advantages and disadvantages were made clear. The distributed fuse method is demonstrated to be further improved by a built-in fuse word driver and a built-in fuse column selector, and fuse analog switches. This enhanced distributed fuse scheme was examined in a fast Bi-CMOS SRAM. A maximun access time of 14 ns and a chip size of 8.8 mm17.4 mm are expected for a 4 Mb Bi-CMOS SRAM in the future.

The paper indicates the importance of suitability assesment in speech synthesis applications. Human factors involved in the use of a synthetic speech are first discussed on the basis of an example of a newspaper company where synthetic speech is extensively used as an aid for proofreading a manuscript. Some findings obtained from perceptual experiments on the subjects' preference for paralinguistic properties of synthetic speech are then described, focusing primarily on the suitability of pitch characteristics, speaker's gender, and speaking rates in the task where subjects are asked to proofread a printed text while listening to the speech. The paper finally claims the need for a flexibile speech synthesis system which helps the users create their own synthetic speech.

Four circuit techniques and a layout design scheme were proposed to realize a 2 Mb VRAM used 0.8 µm technology. They are the enhanced circuit technologies for high speed operation, the functional circuit design and the effective repair schemes for a VRAM, the low power consumption techniques to active and standby mode and a careful layout design scheme realizing high noise immunity. Using these design techniques, a 2 Mb VRAM is suitable for the graphics application of a 5125128 pixels basis screen, with a clear mode of 4.6 GByte/sec and a 4-multi column write mode of 400 MByte/sec, even using the same 0.8 µm technology as the previous VRAM (1 Mb) was realized.

This paper presents a method for verifying safety property of a communication protocol modeled as two extended communicating finite-state machines with two unbounded FIFO channels connecting them. In this method, four types of atomic formulae specifying a condition on a machine and a condition on a sequence of messages in a channel are introduced. A human verifier describes a logical formula which expresses conditions expected to be satisfied by all reachable global states, and a verification system proves that the formula is indeed satisfied by such states (i.e. the formula is an invariant) by induction. If the invariant is never satisfied in any unsafe state, it can be concluded that the protocol it safe. To show the effectiveness of this method, a sample protocol extracted from the data transfer phase of the OSI session protocol was verified by using the verification system.