Statistically improved results of power line radiation (PLR) over Eastern Asia observed at 50 and 60Hz are described in this paper. A total number of 150 orbits, which had been observed from June 1984 to January 1986, by the Japanese scientific satellite OHZORA, are used to detect PLR over Eastern Asia around the Japanese Islands. Depending on the increase in the number of data points, the statistical characteristics of the background noise can be precisely determined by using the improved technique compared with the initial analysis. Statistically reasonable data points are detected as PLR based on the +3 criterion, where is the standard deviation of the background noise. Therefore, the statistical reliability for rejecting the background noise is 99.85%. Then, these detected data are applied to the cause-and-effect test. When the statistically detected data points are placed on the map of Eastern Asia, the points cover Eastern Japan and the east coast of China for 50Hz, and they cover Western Japan for 60 Hz. The maps of the detection ratios and those of the average field strengths indicate the positive correlation with the ground maps of the electric power generation at 50 and 60 Hz. The positive correlation is more clearly seen at 50Hz since the background noise is somewhat weaker than that at 60Hz. This close relationship between the satellite observation and the electric power generation suggests that the detection of PLR is not caused by chance, and that PLR penetrates into the ionosphere and propagates approximately just upward. The decrease of field strength with altitude can be interpreted as the gradual decrease of the refractive index from 400 to 700km. Therefore, the detection ratio and the average field strength with respect to the satellite altitude suggest PLR propagating from the bottom of the ionosphere. According to these observational results, it is concluded that PLR in Eastern Asia is high above the high electric power generating regions over Japan and China, and that the satellite observation is capable of estimating PLR field below the ionosphere. These results are the first direct indication that the PLR field is enhanced over the high electric power generation region, and is penetrating into the ionosphere.

This paper describes a planning method for ATM networks. The method is based on evaluation of two design items, VC routing and VP routing, as well as on consideration of VPI constraints. In the evaluation, VC routing is compared with VP routing in separate case studies undertaken from the point of view of various parameters such as traffic volume, cost function and network scale. The results suggest the vertical relationship between VC and VP levels in optimally designed ATM networks. VC and VP network levels are then studied separately, and design methods are proposed for individual levels. In addition a perturbation method is proposed for the VC and VP routing use, whose optimum is varied as a function of the parameters described above. Evaluation results show the proposed perturbation method provides cost-effective networks.

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.

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.

This paper proposes a new method of phoneme power control for speech synthesis by rule. The innovation of this method lies in its use of the phoneme environment and the relationship between speech power and pitch frequency. First, the permissible threshold (PT) for power modification is measured by subjective experiments using power manipulated speech material. As a result, it is concluded that the PT of power modification is 4.1 dB. This experimental result is significant when discussing power control and gives a criterion for power control accuracy. Next, the relationship between speech power and pitch frequency is analyzed using a very large speech data base. The results show that the relationship between phoneme power and pitch frequency is affected by the kind of phoneme, the adjoining phonemes, rising or falling pitch, and initial or final position in the sentence. Finally, we propose that the phoneme power should be controlled by pitch frequency and phoneme environment. This proposal is implemented in a waveform concatenation type text-to-speech synthesizer. This new method yields an averaged root mean square error between real and estimated speech power of 2.17 dB. This value indicates that 94% of the estimated power values are within the permissible threshold of human perception.

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.

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.

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.

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.

For analyzing the transient electromagnetic fields caused by electrostatic discharge (ESD), a new ESD model is presented here. Numerical calculation is also given to explain the distinctive phenomenon being well-recognized in the ESD event.

We propose an advanced hyper parallel testing method which improves the line-mode test method by adding data inversion registers which we call the Advanced Line-mode Test (ALT). This testing method has the same testing capability as the conventional bit-by-bit and multi-bit test method (MBT), because it enables the application of a high sensitive and practical test patterns under the hyper parallel condition. The testing time for fixed data patterns are reduced by 1/1900 (all-0/1, checker board, and etc.). Moreover, the ALT can be applicable to the continuous patterns (march, walking, and etc.). The ALT improved from the line-mode test with registers and comparators (LTR) is able to applicable to the most test patterns and to reduce the testing time remarkably, and is suitable for the ULSI memories.

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.

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.

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.

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 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.

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.

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.

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.

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.