A novel nonlinear analysis method of high power amplifier instability has been developed. This analysis method deals with a loop oscillation in a closed loop circuit and presents the conditions for oscillation under large-signal operation by taking account of mixing effect of FETs. Applying this analysis to the high power amplifier instability that an output power for the fundamental wave (f0-wave) decreases at some compression point where a half of the fundamental wave (f0/2-wave) is observed, it has been found that this instability is caused by an f0/2 loop oscillation. In addition, it has been verified by analysis and experiment that the oscillation can be removed by employing an isolation resistor in a closed loop circuit.

In the millimeter-wave propagation inside of figure U guideway of maglev, a multipath model using the ray-tracing method is presented. Prediction shows that delay spread is exceedingly small and high speed data transmission more than 100Mbps is possible without an equalizer.

We have designed and implemented a simple yet powerful declarative synchronization mechanism for a paralle object-oriented computation model. The mechanism allows the user to control multiple message reception, specify the order of message reception, lock an invocation, and specify relations as invocation constraints. It has been included in a parallel object-oriented language, called A-NETL. The compiler and operating system have been developed on a total architecture, A-NET (Actors NETwork). The experimental results show that (i) the mechanism allows the user to model asynchronous events naturally, without losing the integrity of described programs; (ii) the replacement of the mechanism with the user's code requires tedious descriptions, but gains little performance enhancement, and certainly loses program readability and integrity; (iii) the mechanism allows the user to shift synchronous programs to asynchronous ones, with a scalable reduction of execution times: an average 20.6% for 6 to 17 objects and 46.1% for 65 objects. These prove the effectiveness of the proposed synchronization mechanism.

An experimental study on propagation characteristics in a room of a modern office building using a 57.5 GHz carrier is described. The objective of the study is to provide initial information about the path-loss and fading characteristics for millimeter-wave wireless indoor communication systems. The area of the floor of the room is about 90 m2 and the room was almost empty with no furniture except the experimental apparatus. A wide-beam scalar feed horn transmission antenna and an omnidirectional receiving antenna were used and set to be with the same height. The measured area was divided into many "squares" to characterize the propagation. The variation of square transmission loss was calculated, and the regions where the maximum square transmission loss was observed were pointed out. It became clear that the distance power law for square transmission loss was approximately of the order d-2 along the direct path. Rician distributions for fast fading were derived in almost all squares of this room. The spectra for the fast fading envelope were also derived. From the propagation point of view, the results show that no extreme difficulty seems to exist in developing wireless communication systems in the 60 GHz band.

A bottleneck identification methodology is proposed for the performance-oriented design of shared-bus multiprocessors, which are composed of several major subsystems (e.g. off-chip cache, bus, memory, I/O). A subsystem with the longest access time per instruction is the one that limits processor performance and creates a bottleneck to the system. The methodology also facilitates further refined analysis on the access time of the bottleneck subsystem to help identify the causes of the bottleneck. Example performance model of a particular shared-bus multiprocessor architecture with separate address bus and data bus is developed to illustrate the key idea of the bottleneck identification methodology. Accessing conflicts in subsystems and DMA transfers are also considered in the model.

Improvement of the convergence characteristics of the NLMS algorithm has received attention in the area of adaptive filtering. A new variable stepsize NLMS method, in which the stepsize is updated optimally by using variances of the measured error signal and the estimated noise, is proposed. The optimal control equation of the stepsize has been derived from a convergence characteristic approximation. A new condition to judge convergence is introduced in this paper to ensure the fastest initial convergence speed by providing precise timing to start estimating noise level. And further, some adaptive smoothing devices have been added into the ADF to overcome the saturation problem of the identification error caused by some random deviations. By the simulation, The initial convergence speed and the identification error in precise identification mode is improved significantly by more precise adjustment of stepsize without increasing in computational cost. The results are the best ever reported performanced. This variable stepsize NLMS-ADF also shows good effectiveness even in severe conditions, such as noisy or fast changing circumstances.

An error correction/detection decoding scheme of binary Hamming codes is proposed. Error correction is performed by algebraic decoding and then error detection is performed by simple likelihood ratio testing. The proposed scheme reduces the probability of undetected decoding error in comparison with conventional error correction scheme and increases throughjput in comparison with conventional error detection scheme.

A description language for matrix and vector expressions and its compiler for DSPs are shown. They provide both a user-friendly programming environment and efficient codes. In order to increase throughput and to reduce amount of methods based on mathematical laws are introduced. A method to decide the matrix and vector storage location suitable for processing on DSP is also proposed.

Data flow is a paradigm for concurrent computations in which a collection of parallel processes communicate asynchronously. For nondeterministic data flow networks many semantic models have been defined, however, it is complex to reason about the semantics of a network. In this paper, we introduce a transformation between data flow networks and the LOTOS specification language to make available theories and tools developed for process algebras for the semantic analysis based on traces of the networks. The transformation does not establish a one-to-one mapping between the traces of a data flow network and the LOTOS specification, but maps each network in a specification which usually contains more traces. The obtained system specification has the same set of traces as the corresponding network if they are finite, otherwise also non fair traces are included. Formal analysis and verification methods can still be applied to prove properties of the original data flow network, allowing in case of networks with finite traces to prove also network equivalence.

This paper deals with a high-speed digital circuit for discrete cosine transform (DCT). We propose a new algorithm that reduces the number of calculations for partial sum-of-products in the DCT and synthesize the small gate depth circuit of DCT by using carry-propagation-free adders based on redundant binary {1,0,1} representation. The gate depth is only half to one third that of the conventional algorithms with the same number of gates.

This paper discusses advanced wireless communication technologies for achieving future high-speed mobile radios. Mainly, five technical fields are considered, that is, multi-level modulation for transmitting high-capacity information signal, advanced adaptive wireless system flexibly changing modulation level, symbol rate and traffic according to fading conditions, adaptive multicarrier system transmitting multimedia signals by changing the number of carrier according to the capacity of the signals, new CDMA techniques for mapping different bit rate services onto the same allocated bandwidth at the same time, and optical-linked microcellular communication system with millimeter wave air interface.

This paper presents an improved neural network for channel assignment problems in cellular mobile communication systems in the new co-channel interference model. Sengoku et al. first proposed the neural network for the same problem, which can find solutions only in small size cellular systems with up to 40 cells in our simulations. For the practical use in the next generation's cellular systems, the performance of our improved neural network is verified by large size cellular systems with up to 500 cells. The newly defined energy function and the motion equation with two heuristics in our neural network achieve the goal of finding optimum or near-optimum solutions in a nearly constant time.

A generalized surface scattering radar equation for a near-nadir-looking pencil beam radar, which covers both beam-limited and pulse-limited regions, is derived. This equation is a generalization of the commonly used nadir-pointing beam-limited radar equation taking both antenna beam and pulse wave form weighting functions into account, and is convenient for the calculation of radar received power and scattering cross-section of the surface.

A down sampling technique was applied to signal processing of fiber optic gyroscopes with optical phase modulation. The technique shifts the frequency spectrum of the gyroscopic signal down to low frequencies, and lowers the speed requirements for analog-to-digital (A/D) conversion and numerical operations. A single-chip digital signal processor (DSP) with a built-in A/D converter and timers was used to demonstrate the proposed technique. The DSP internally generated a phase modulation signal and sampling trigger timing. The reference signals for digital lock-in discrimination of gyroscopic spectrum are generated by using an external binary counter, and their phases were adjusted optimally by DSP software. The DSP compensated for fluctuations in laser source intensity and phase modulation index, using the signal spectrum extracted, and linearized the gyroscopic response. The measured resolution of rotation detection was 0.9 deg/s (with a full scale of 100 deg/s) and it agreed with the resolution in A/D conversion.

This paper describes modifications to a previously proposed 8-kb/s 4-ms-delay CELP speech coding algorithm with a view to improving the speech quality while maintaining low delay and only moderately increasing complexity. The modifications are intended to improve the effectiveness of interframe pitch lag prediction and the sub-optimality level of the excitation coding to the backward adapted synthesis filter by using delayed decision and joint optimization techniques. Results of subjective listening tests using Japanese speech indicate that the coded speech quality is significantly superior to that of the 8-kb/s VSELP coder which has a 20-ms delay. A method that reduces the computational complexity of closed-loop 3-tap pitch prediction with no perceptible degradation in speech quality is proposed, based on representing the pitch-tap vector as the product of a scalar pitch gain and a normalized shape codevector.

A multi-dimensional shaping scheme based on multi-level Maximum Average Weight (MAW)-codes is presented. One can reduce the average energy of transmitted signal, by using low energy signal points more frequently than high energy ones. The proposed scheme employs a multi-dimensional region of 2,4,6 and 8 dimensions; these regions are selected using a multi-level MAW-code. A multi-level MAW-code is a q-ary code and has unequal probability of the occurrence of a symbol. The scheme can achieve a shaping gain of 0.6-1.0 dB with small constellation expansion ratio and peak to average energy ratio. This scheme is based on a two-level table look up algorithm. Therefore, the less complexity of encoding/decoding can be realized.

This paper describes a system that can enchance the speech quality degradation due to severe band limitation during speech transmission. We have already proposed a spectrum widening method that utilizes aliasing in sampling rate conversion and digital filtering for spectrum shaping. This paper proposes a new method that offers improved performance in terms of the spectrum distortion characteristics. Implementation procedures are clarified, and its performance is discussed. The proposed method can effectively enhance speech quality.

This paper describes a spatial and temporal multipath channel model which is useful in array antenna environments for mobile radio communications. From this model, a no distortion criterion, that is an extension of the Nyquist criterion, is derived for equalization in both spatial and temporal domains. An adaptive tapped-delay-line (TDL) array antenna is used as a tool for equalization in both spatial and temporal domains. Several criterion for such spatial and temporal equalization such as ZF (Zero Forcing) and MSE (Mean Square Error), are available to update the weights and tap coefficients. In this paper, we discuss the optimum weights based on the ZF criterion in both spatial and temporal domains. Since the ZF criterion satisfies the Nyquist criterion in case of noise free, this paper applies the ZF criterion for the spatial and temporal equalization as a simple case. The Z transform is applied to represent the spatial and temporal model of the multipath channel and to derive the optimal weights of the TDL array antenna. However, in some cases the optimal antenna weights cannot be decided uniquely. Therefore, the effect on the equalization errors due to a finite number of antenna elements and tap coefficients can be shown numerically by computer simulations.

A new variable step size Least Mean Square (LMS) FIR adaptive filter algorithm (VSS-CC) is proposed. In the VSS-CC algorithm the step size adjustment (α) is controlled by using the correlation between the output error (e(n)) and the adaptive filter output ((n)). At small times, e(n) and (n) are correlated which will cause a large α providing faster tracking. When the algorithm converges, the correlation will result in a small size α to yield smaller misadjustments. Computer simulations show that the proposed VSS-CC algori thm achieves a better Echo Return Loss Enhancemen (ERLE) than a conventional NLMS Algorithm. The VSS-CC algorithm was also compared with another variable step algorithm, achieving the VSS-CC a better ERLE when the additive noise is incremented.

This paper condiders a problem of logecal configuration in reconfigurable VCDN (Virtual Circuit Data Networks) which is analyzed through a mimimax approach, and its objective is to minimize the largest delay on any logical link, measured in both queueing delay and propagation delay. The problem is formulated as a 0/1 mixed integer programming and analyzed by decomposing it into two subproblems, called routing and dimensioning problems, for which an efficient hauristic algorithm is proposed in an iterating process made beween the two subproblems for solution improvement. The algorithm is tested for its performance eveluation.