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.

We propose a smart design methodology for advanced ULSI memories to reduce the turn around time(TAT) for circuit revisions with no area penalty. This methodology was executed by distributing extra gate-arrays, which were composed of the n-channel and p-channel transistors, under the power line and the signal line. This method was applied to the development of a 16 Mb DRAM with double metal wiring. The design TAT can be reduced to 1/8 using 1500 gates. This design methodology has been confirmed to be very effective.

A new system for speech synthesis by concatenating waveforms selected from a dictionary is described. The dictionary is constructed from a two-hour speech that includes isolated words and sentences uttered by one male speaker, and contains over 45,000 entries which are identified by their average pitch, dynamic pitch parameter which represents micro pitch structure in a segment, duration and average amplitude. Phoneme duration is set according to phoneme environment, and phoneme power is controlled, by both pitch frequency and phoneme environment. Tests show the average errors in vowel duration and consonant duration are 28.8 ms and 16.8 ms respectively, and the vowel power average error is 2.9 dB. The pitch frequency patterns are calculated according to a conventional model in which the accent component is abbed to a gross phrase component. Set a phoneme string and prosody information, the optimum waveforms are selected from the dictionary by matching their attributes with the given phonetic and prosodic information. A waveform selection function, which has two terms corresponding to prosody and phonological coincidence between rule-set values and waveform values from the dictionary, is proposed. The weight coefficients used in the selection function are determined through subjective hearing tests. The selected waveform segments are then modified in waveform domain to further adjust for the desired prosody. A pitch frequency modification method based on pitch synchronous overlap-add technique is introduced into the system. Lastly, the waveforms are interpolated between voiced waveforms to avoid abrupt changes in voice spectrum and waveform shape. An absolute evaluation test of five grades is performed to the synthesized voice and the mean of the score is 3.1, which is over "good," and while the original speaker quality is retained.

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%.

Text-to-speech synthesis (TTS) is currently one of the most important media conversion techniques. In this paper, we describe a Japanese TTS card developed for constructing a personal-computer-based multimedia platform, and a TTS software package developed for a workstation-based multimedia platform. Some applications of this hardware and software are also discussed. The TTS consists of a linguistic processing stage for converting text into phonetic and prosodic information, and a speech processing stage for producing speech from the phonetic and prosodic symbols. The linguistic processing stage uses morphological analysis, rewriting rules for accent movement and pause insertion, and other techniques to impart correct accentuation and a natural-sounding intonation to the synthesized speech. The speech processing stage employs the cepstrum method with consonant-vowel (CV) syllables as the synthesis unit to achieve clear and smooth synthesized speech. All of the processing for converting Japanese text (consisting of mixed Japanese Kanji and Kana characters) to synthesized speech is done internally on the TTS card. This allows the card to be used widely in various applications, including electronic mail and telephone service systems without placing any processing burden on the personal computer. The TTS software was used for an E-mail reading tool on a workstation.

The concept of functional memory was proposed over nearly four decades ago. However, the actually usable products have not appeared until the 1980s instead of the long history of development. Functional memory is classified into three categories; there are a general functional memory, a processing element array with small size memory and a special purpose memory. Today a majority of functional memory is an associative memory or a content addressable memory (CAM) and a special purpose memory based on CAM. Due to advances in fablication capability,the capacity of CAM LSI has increased over 100 K bits. A general purpose CAM was developed based on SRAM cell and DRAM cell, respectively. The typical CAM LSI of both types, 20 K bits SRAM based CAM and 288 K bits DRAM based CAM, are introduced. DRAM based CAM is attractive for the large capacity. A parallel processor architecture based on CAM cell is proposed which is called a Functional Memory Type Parallel Processor (FMPP). The basic feature is a dual character of a higher performance CAM and a tiny processor array. It can perform a highly parallel operation to the stored data.

Until now, in a communication system which deals with multiple processes, system behavior has been described by a fixed number of processes. The state reachability problem for specified processes was generally deliberated within a pre-defined number of processes, and was analyzed by essentially searching for all possible behaviors. However, in a system whose number of processes is arbitrary, a given state which is not reachable in some situations which consists of a small number of processes might be reachable in another situation which consists of a larger number of processes. This article discusses the above problem, assuming that the behavior of a system is described by an arbitrary number of processes. After discussing the relationship between our model and the Petri net model, we clarify the properties between the set of reachable states and the number of processes involved in the system, and show an algorithm to obtain a sufficient number of processes for resolving the reachability problem.

Queueing problems are investigated for very wide classes of input traffic and service time models to obtain good loss probability and waiting time probability approximation. The proposed approximation is based on the fundamental recursion formula and the Chernoff bound technique, both of which requires no particular assumption for the stochastic nature of input traffic and service time, such as renewal or markovian properties. The only essential assumption is stationarity. We see that the accuracy of the obtained approximation is confirmed by comparison with computer simulation. There are a number of advantages of the proposed method of approximation when we apply it to network capacity design or path accommodation design problems. First, the proposed method has the advantage of applying to multi-media traffic. In the ATM network, a variety of bursty or non-bursty cell traffic exist and are superposed, so some unified analysis methodology is required without depending each traffic's characteristics. Since our method assumes only the stationarity of input and service process, it is applicable to arbitrary types of cell streams. Further, this approach can be used for the unexpected future traffic models. The second advantage in application is that the proposed probability approximation requires only small amount of computational complexity. Because of the use of the Chernoff bound technique, the convolution of every traffic's probability density fnuction is replaced by the product of probability generating functions. Hence, the proposed method provides a fast algorithm for, say, the call admission control problem. Third, it has the advantage of accuracy. In this paper, we applied the approxmation to the cases of homogeneous CBR traffic, non-homogeneous CBR traffic, M/D/1, AR(1)/D/1, M/M/1 and D/M/1. In all cases, the approximating values have enough accuracy for the exact values or computer simulation results from low traffic load to high load. Moreover, in all cases of the numerical comparison, our approximations are upper bounds of the real values. This is very important for the sake of conservative network design.

We analyze frame rates in low bit rate video coding and show that an optimal frame rate can be theoretically obtained. In low bit rate video coding the frame rate is usually forced to be decreased for reducing the total amount of coded information. The choice of frame rate, however, has a great effect on the picture quality in a trade-off relation between coded picture quality and motion smoothness. It is known from experience that in order to achieve an optimum balance between these two factors, a frame rate has to be selected which is appropriate for the coding scheme, property of the video sequences and coding bit rate. A theoretical analysis, however, on the existence of an optimal frame rate and how the optimal frame rate would be expressed has not been performed. In this paper, coding distortion measured by mean square error is analyzed by using video signal models such as a rate-distortion function for coded frames and inter-frame correlation coefficients for non-coded frames. Overall picture quality taking account of coded picture quality and motion smoothness simultaneously is expressed as a function of frame rate. This analysis shows that the optimum frame rate can be uniquely specified. The maximum frame rate is optimal when the coding bit rate is higher than a certain value for a given video scene, while a frame rate less than the maximum is optimal otherwise. The result of the theoretical analysis is compared with the results of computer simulation. In addition, the relation between this analysis and a subjective evaluation is described. From both comparisons this theoretical analysis can be justified as an effective scheme to indicate the optimal frame rate, and it shows the possibility of improving picture quality by selecting frame rate adaptively.

In our preceding paper, I-ISMA (Idle Signal Multiple Access for Integrated services), a combination of ISMA and time reservation technique, was proposed to transmit an integrated voice and data traffic in third generation wireless communication networks. There, the channel capacity of I-ISMA was evaluated by the static analysis. To fully estimate performance of contention-based channel access protocols, however, we also need dynamic analysis to evaluate stability, delay, etc. Particularly, in systems concerning real-time voice transmission, delay is one of the most important performance measures. A six-mode model to describe an I-ISMA system is set up. With some assumptions for simplification, the dynamic behavior of the system is approximated by a Markov process so that the EPA (Equilibrium Point Analysis), a fluid approximation method, can be applied to the analysis. Then, numerical and simulation results are obtained for some examples. By means of the same analysis method and under the same conditions, the performance of PRMA is evaluated and compared briefly with that of I-ISMA.

This paper describes a novel method to calculate the fields scattered by a polyhedron structure for an incident plane wave. In this method, the fields diffracted by an edge are calculated using the equivalent edge currents which are separated into components dependent on each of the two surfaces which form the edge. The separated equivalent edge currents are based on the Geometrical Theory of Diffraction (GTD). Using this Separated Equivalent Edge Current Method (SEECM) , fields scattered by a polyhedron structure can be calculated without special treatment of the singularity in the diffraction coefficient. This method can be also applied successfully to structures with convex surfaces by modeling them as polyhedron structures.

This paper proposes a new scheme for concatenative speech synthesis to improve the speech segment selection procedure. The proposed scheme selects a segment sequence for concatenation by minimizing acoustic distortions between the selected segment and the desired spectrum for the target without the use of heuristics. Four types of distortion, a) the spectral prototypicality of a segment, b) the spectral difference between the source and target contexts, c) the degradation resulting from concatenation of phonemes, and d) the acoustic discontinuity between the concatenated segments, are formulated as acoustic quantities, and used as measures for minimization. A search method for selecting segments from a large speech database is also descrided. In this method, a three-step optimization using dynamic programming is used to minimize the four types of distortion. A perceptual test shows that this proposed segment selection method with minimum distortion criteria produces high quality synthesized speech, and that contextual spectral difference and acoustic discontinuity at the segment boundary are important measures for improving the quality.

Numerical analysis of the electromagnetic radiation from conducting surface structures is concerned. The method of moments is discussed with the surface-patch modeling in which the surface quantities, i.e. the current, charge and impedance are directly introduced and with the wire-grid modeling in which the surface quantities are approximated by the filamentary traces. The crucial element to a numerical advantage of the wire-grid modeling lies in the simplicity of its mathematical involvements that should be traded for the uncertainties in the construction of the model. The surface-patch techniques are generally not only clear and straightforward but also more reliable than the wire-grid modeling for the computation of the surface quantities. In this work, we bring about a comparative discussion of the two approaches while the analysis of a built-in planar antenna is reported. For the purpose of the comparison, the same electric field integral equation and the Galerkin's procedure with the linear expansion/testing functions are used for both the wire-grid and surface-patch modeling.

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 paper describes a bitline control circuit and redundancy technique for high-density dynamic content addressable memories (CAMs). The proposed bitline control circuit can efficiently manage a dynamic CAM cell accompanied by complex operations; that is, a refresh operation, a masked search operation, and partial writing, in addition to normal read/write/search operations. By adding a small supplementary circuit to the bitline control circuit, a circuit scheme with redundancy which prevents disabled column circuits from affecting a match operation can also be obtained. These circuit technologies achieve higher-density dynamic CAMs than conventional static CAMs. These technologies have been successfully applied to a 288-kbit CAM with a typical cycle time of 150 ns.

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.

Space-time tradeoff is a very fundamental issue to design a fault-tolerant real-time (called responsive) system. Routing a message in large computer networks is efficient when each node knows the full topology of the whole network. However, in the hierarchical routing schemes, no node knows the full topology. In this paper, a tradeoff between an optimality of path length (message delay: time) and the amount of topology information (routing table size: space) in each node is presented. The schemes to be analyzed include K-scheme (by Kamoun and Kleinrock), G-scheme (by Garcia and Shacham), and I-scheme (by authors). The analysis is performed by simulation experiments. The results show that, with respect to average path length, I-scheme is superior to both K-scheme and G-scheme, and that K-scheme is better than G-scheme. Additionally, an average path length in I-scheme is about 20% longer than the optimal path length. On the other hand, for the routing table size, three schemes are ranked in reverse direction. However, with respect to the order of size of routing table, the schemes have the same complexity O (log n) where n is the number of nodes in a network.

The emerging discipline of responsive systems demands fault-tolerant and real-time performance in uniprocessor, parallel, and distributed computing environments. The new proposal for responsiveness measure is presented, followed by an introduction of a model for responsive computing. The model, called CONCORDS (CONsensus/COmputation for Responsive Distributed Systems), is based on the integration of various forms of consensus and computation (progress or recovery). The consensus tasks include clock synchronization, diagnosis, checkpointing scheduling and resource allocation.

Typical processes controlled by hard real-time computer systems undergo several, mutually exclusive modes of operation. By deterministically switching among a number of static schedules, a pre run-time scheduled system is able to adapt to changing environmental situations. This paper presents concepts for specification of mode changes, construction of static schedules for modes and transitions, and timely run-time execution of mode changes. We propose concepts for mode changes in the context pre run-time scheduled hard real-time systems. While MARS is used to illustrate the concepts' application, they are applicable to a variety of systems. Our methods adhere closely to the ones established for single modes. By decomposing the system into a set of disjoint modes, the design process and its comprehension are facilitated, testing efforts are reduced significantly, and solutions are enabled which do not exist if all system activities of all modes are combined into a single schedule.

This paper proposes a group-to-group communications algorithm that can extend the range of distributed systems where we can achieve active replication fault-tolerance to partner model distributed systems, in which all processes communicate with each other on an equal footing. Active replication approach, in which all replicated processes are active, can achieve fault-tolerance with low overhead because checkhpoint setting and rollback are not required for recovery from process failure. This algorithm guarantees that each replicated process in a process group has the same execution history and that communications between process groups keeps consistency even in the presence of process failure and message loss. The number of control messages that must be transmitted between processes for a communication between process groups is only a linear order of the number of replicated processes in each process group. Furthemore, this algorithm reduces the overhead for reconfiguration of a process group by keeping process failure and recovery information local to each process group.