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 proposes a symbol rate controlled adaptive modulation/TDMA/TDD for future wireless personal communication systems. The proposed system controls the symbol rate according to the channel conditions to achieve wide dynamic range of the modulation parameter control as well as to improve the delay spread immunity. The main purpose of the proposed system is to increase the data throughput with keeping a certain transmission quality, especially in frequency selective fading environments. For this purpose, the proposed system predicts the C/N0 (carrier power-to-noise spectral density ratio) and the delay spread separately, and selects the optimum symbol rate that gives the maximum bit rate within a given bandwidth satisfying the required BER. The simulated results show that the proposed system can achieve higher transmission quality in comparison with the fixed symbol rate transmission system in both flat Rayleigh and frequency selective fading environments. The results also show that the proposed system is very effective to achieve higher bit rate transmission in frequency selective fading environments.

This paper proposes a high throughput small latent IP packet delivery architecture using ATM technology in a large scaled internet. Data-link network segments, including ATM network segments, are interconnected through routers. A connection oriented IP packet delivery will be provided by IP (including both IPv4 and IPv6) with a certain resource reservation protocol (e.g. RSVP). When the router attached to ATM network segment has a mapping function between the flow-ID (e.g. in the SIPP header) and the VPI/VCI value, the small latent connection oriented IP forwarding can be provided. Also, when the router has cell-relaying functionality, the small latent connectionless IP forwarding can be provided, even in IPv4. The source router, where the source end-station belongs to, will be able to transfer the connectionless IP packet to the destination router, where the destination end-station belongs to, through the concatenated ATM connections (ATM-VCCs) without any ATM-VCC termination point. When all of the network segments are ATM-LAN, the proposed architecture can accommodate about up to 222 (4106) end-stations with two network layer processing points. And when the network is scaled up hierarchally, we can accommodate larger number of end-stations. For example, we can accommodate 1015 end-stations by a three layered network. Then the maximum number of actual network layer processing points between source and destination end-stations can be ten. Here, 1015 is the maximum number of end-stations in ISDN and also it is the target number of accommodated end-stations for IPv6.

A signal-to-noise enhancer with a bandwidth that is six times as wide as that of the conventional type is presented. A new circuit construction, the combination of two MSSW filters which have the same insertion loss in the broadband and two 90 hybrids, is effective to remarkably extend the bandwidth. The enhancement of the enhancer amounts to 20 dB in the operating frequency range of 1.9 GHz150 MHz in 0 to 60 degrees centigrade. This enhancer has accomplished FM threshold extension because the S/N is improved by 1 to 7 dB below the C/N of 9 dB. It was demonstrated that this new enhancer is effective for noise reduction in practical DBS reception.

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.

The purpose of this research is to analyze production rules with coupling modes in active databases and to exploit an assistant system for rule programming. Each production rule is a specification including an event, a condition, and an action. The action is automatically executed whenever the event occurs and the condition is satisfied. Coupling modes are useful to control execution order of transactions. For example, a transaction for consistency check should be executed after transactions for update. An active database, which is a database with production rules, can spontaneously update database states and check their consistency. Production rules provide a powerful mechanism for knowledge-bases. However it is very difficult in general to predict how a set of production rules will behave because of cascading rule triggers, concurrency, and so on. We are attempting to adopt a process algebra as a basic tool to analyze production rules. In order to describe and analyze concurrent and communicating systems, process algebras such as CCS, CSP, ACP, and π-calculus, are well known. However there are some difficulties to apply existing process algebras to analysis of production rules in growing process trees by process creation. In this paper we propose a process algebra named CCSPR (a Calculus of Communicating Systems with Production Rules), Which is an extension of CCS. An advantage of CCSPR is to syntactically describe growing process trees. Therefore, production rules can be appropriately analyzed in CCSPR. After giving definitions and properties of CCSPR, we show an example of analysis of production rules in CCSPR.

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 transmission power control technique on a TDD-CDMA/TDMA system for wireless multimedia networks is proposed. The assumed network connects mobile terminals to a node of an ATM based high speed LAN through a radio central unit. Only human interface facilities are implemented into the terminal so that users access integrated services through the node of the network. The uplink (from a mobile terminal to a radio central unit) employs a CDMA scheme to transmit human interface signals (2.4kbit/s) and the downlink employs a TDMA scheme to transmit display interface signals (24 Mbit/s). Both the CDMA and the TDMA signals occupy the same frequency band. To mitigate bit error rate degradation due to the fading, the radio central unit estimates the impulse response of the channel from the received CDMA signals and controls the transmission power of the TDMA signals to compensate the fading attenuation. The bit error rate performance of the downlink with the proposed transmission power control is theoretically analyzed under several fading conditions. Numerical results using the Nakagami-m fading model and recent propagation measurements show that the proposed power control technique compensates the fading attenuation and improves the bit error rate performances. The bit error rate of the downlink is reduced from 10-2 to 10-5 at the symbol SNR of 20dB by employing the proposed transmission power control, which is less sensitive to the severity of the fading. Furthermore, the proposed transmission power control is implemented without increasing the terminal complexity because all the processing on the power control of the downlink is carried out only in the radio central unit.

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.

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.

Distributed computing over a network of workstations continues to be an illusive goal. Its main obstacle is the delay penalty due to network protocol and OS overhead. We present in this paper a low level hardware supported scheme for managing distributed shared memory (DSM), as an underlying paradigm for distributed computing. The proposed DSM is novel in that it employs a two-tier paging scheme that reduces the probability of false sharing and facilitates an efficient hardware implementation. The scheme employs a standard OS page and divides it into fixed smaller memory units called paragraphs, similar to cache lines. This scheme manages the shared data regions only, while other regions are handled by the OS in the standard manner without modification. A hardware extension of a traditional MMU, namely Distributed MMU or DMMU, is introduced to support the DSM. Shared memory coherency is maintained through a write-invalidate protocol. An analytical model is built to evaluate the system sensitivity to various parameters and to assess its performance.

This paper describes the multiple access performance of parallel combinatory spread spectrum (PC/SS) communication systems in nonfading and Rayleigh fading multipath channels. The PC/SS systems can provide the high-speed data transmission capability by transmitting multiple pseudo-noise sequences out of a pre-assigned sequence set. The performance is evaluated in terms of average bit error rate (BER) by numerical computation. In nonfading white gaussian channel, the PC/SS systems are superior to conventional direct sequence spread spectrum (DS/SS) systems under the identical spreading factor condition. In Rayleigh fading channel, the performance of the PC/SS system without diversity is poorer than that of the DS/SS system. By including the explicit and implicit diversity, the performance of the PC/SS system becomes better than that of conventional DS/SS systems. A longer spreading sequence is assignable to a PC/SS system having the spreading factor equal to that in the conventional DS/SS system. Hence, the error control coding is easily. It is found that the PC/SS systems including diversity and Reed-Solomon coding improves the multiple access performance.

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.

The back-propagation algorithm has been applied to many fields, and has shown large capability of neural networks. Many people use the back-propagation algorithm together with a momentum term to accelerate its convergence. However, in spite of the importance for theoretical studies, theoretical background of a momentum term has been unknown so far. First, this paper explains clearly the theoretical origin of a momentum term in the back-propagation algorithm for both a batch mode learning and a pattern-by-pattern learning. We will prove that the back-propagation algorithm having a momentum term can be derived through the following two assumptions: 1) The cost function is Enαn-µEµ, where Eµ is the summation of squared error at the output layer at the µth learning time and a is the momentum coefficient. 2) The latest weights are assumed in calculating the cost function En. Next, we derive a simple relationship between momentum, learning rate, and learning speed and then further discussion is made with computer simulation.

To support high bit rate and high quality indoor radio communication systems, we have to solve intersymbol interference (ISI) problem caused by frequency-selective fading. Recently multi-carrier modulation technique is considered to be one of the effective methods for this problem. In this paper we propose Partial Frequency ARQ (Automatic Repeat reQuest) system which can achieve effective ARQ scheme for multi-carrier packet communication. This system operates partial retransmission of erroneous power faded packets, and it is superior to the traditional ARQ systems. Furthermore two different protocols are examined for this system: Static Carrier Assignment (SCA) and Dynamic Carrier Assignment (DCA). By computer simulation we found that DCA method can achieve better performance than SCA in terms of both throughput and packet transmission delay.

The scattering characteristics of a waveguide T-junction with an inductive post are analyzed by the port reflection coefficient method (PRCM), combined with the mode-matching technique. Variation behaviors of the scattering parameters are provided as a function of the operating frequency and the dimensions of the junction. The results are helpful for the design of power dividers using this type of T-junction configuration.

Standard Site Method (SSM) is theoretically analyzed using matrix representations to examine its validity and develop an improved method. The analysis reveals that the SSM yields an antenna factor specifically related to the effective load impedance presented by the cable and associated devices which are disconnected from the antenna during the SSM site attenuation measurements. Therefore, an additional conversion is required to determine the desired antenna factor under actual load conditions. It is also concluded that the SSM is not applicable to antennas having height-dependent antenna factors. In addition, the SSM correction factors are found to be theoretically inappropriate. Uncertainty of the antenna factor obtained using the SSM is discussed and the required antenna separation distance is investigated. To improve the existing SSM, it is proposed that both transmitting and receiving antennas are placed at the same height during the site attenuation measurements. Experiments exhibit the superiority of the improved method.

This paper describes a flexible point-to-multipoint access protocol for the fiber-optic passive double star (PDS) system. To provide various types of services, and permit flexibility in changing transport capacity, a time division multiple access (TDMA) scheme for the PDS system is considered. Dynamic time slot multiplexing based on TDMA is proposed to provide required time slots efficiently according to service changes. The effectiveness of dynamic time slot multiplexing is calculated and compared to fixed time slot multiplexing for telephony services. A TCM/TDMA frame structure and an access protocol enabling dynamic time slot multiplexing are proposed. ONU bandwidth is dynamically assigned by using a set of pointers. The ONU access protocol causes no interruption to operating ONUs on the same PDS system during the configuration or reconfiguration of an ONU. The access time is analyzed to estimate the performance of the access protocol. The probability density of access time is calculated for the number of ONUs connected. The calculation results indicate that a PDS system can accommodate up to around 60 ONUs within the maximum access time specified by ITU-T. The experimental results also agree fairly well with the theoretical values.

Current testing has been proposed as an alternative technique for testing fully CMOS digital LSIs. Current testing has higher fault coverage than conventional stuck-at fault (SAF) testing and is more economical because it detects a wide range of faults and requires fewer test vectors than does SAF testing. We have proposed a current testing that measures the integral of the power supply current (IDD) during one clock period including the switching current. Since this method cannot be affected by the switching current, it can be used to test an LSI operating at a relatively high clock freuqnecy. This paper presents an improved current testing method for CMOS digital and analog LSIs. The method uses two current values (i.e., an upper limit and a lower limit) and judges the circuit under test to be faulty if the measured IDD is outside these limits. The proposed current testing is evaluated here for some kinds of faults (e.g., the bridging fault and the breaking fault) in digital and mixed-signal LSIs, and its efficiency of the current testing using SPICE3.