This letter introduces an alternative adaptive beamforming with the total computational load of about O(3N) where N denotes the number of antenna elements. The proposed technique finds a weight vector that maximizes the received signal power at the array output by searching for the suboptimal phase of each weight in a multipath fading CDMA mobile communication environment.

Iterative decoding is a key feature of turbo code and each decoding results in additional power consumption of the decoder and decoding delay. In this letter, we propose an effective stop criterion based on the Gaussian assumption at the decoder output. Simulation results show that the proposed method can dynamically stop the iterative process with a negligible degradation of the error performance.

This paper proposes a design method of variable IIR digital filters based on balanced realizations and minimum round-off noise realizations of digital filters. Highly accurate variable digital filters are easily derived by the proposed method. The coefficient matrices of both realizations of second-order digital filters are obtained directly from prototype realizations. The filter coefficients of variable digital filters can be obtained by frequency transformations to the realizations. The filter coefficients are presented as truncated Taylor series for the purpose of reducing a number of calculations to tune the coefficients. However the proposed filters have highly accurate variable characteristics against the coefficient truncation since balanced realizations and minimum round-off noise realizations have very low coefficient sensitivities, which are invariant under the frequency transformations. Moreover, the dynamic ranges of the proposed filters are almost constant against the frequency transformations. Numerical examples show the effectiveness of the variable digital filters designed by the proposed method.

CRL (Communications Research Laboratory, Independent Administrative Institution Japan) is developing a road to vehicle multiple-service communication system based on RoF (Radio over Fiber) technology in a millimeter-wave frequency region of 36-37 GHz. In the experimental system, vehicle can receive three wireless services such as PHS (Personal Handy-phone System), ETC (Electronic Toll Collection system), SB (Satellite Broadcasting). In this paper, the system concept and experimental system configuration are introduced. Furthermore, SDR (Software Defined Radio) mobile terminal technology is mentioned and a new concept for a next generation mobile communication network system based on RoF is proposed.

The adaptive system design by 16-bit fixed point processing enables to employ an inexpensive digital signal processor (DSP). The narrow dynamic range of such 16 bits, however, does not guarantee the same performance that is confirmed beforehand by computer simulations. A cause of degrading the performance originates in the operation halving the word length doubled by multiplication. This operation rounds off small signals staying in the lower half of the doubled word length to zero. This problem can be solved by limiting the multiplier to only its sign () like the signed regressor algorithm, named 'bi-quantized-x' algorithm in this paper, for the convenience mentioned below. This paper first derives the equation describing the convergence property provided by a type of signed regressor algorithms, the bi-quantized-x normalized least mean square (NLMS) algorithm, and then formulates its convergence condition and the step size maximizing the convergence rate. This paper second presents a technique to improve the convergence property. The bi-qiantized-x NLMS algorithm quantizes the reference signal to 1 according to the sign of the reference signal, whereas the technique moreover assigns zero to the reference signal whose amplitude is less than a predetermined level. This paper explains the principle that the 'tri-qunatized-x' NLMS algorithm employing the technique can improve the convergence property, and confirms the improvement effect by computer simulations.

An algorithm for blind identification of multichannel (single-input and multiple-output) FIR systems is proposed. The proposed algorithm is based on subspace approach to blind identification, which requires so-called noise space spanned by some eigenvectors of correlation matrices of observations. It is shown that a subspace of the noise space can be obtained by one-step scalar-valued linear prediction and then the subspace is sufficient for blind identification. To acquire the subspace, the proposed algorithm utilizes one-step scalar-valued linear prediction in place of a singular- (or eigen-) value decomposition and hence it is computationally efficient. Computer simulations are presented to compare the proposed algorithm with the original one.

This paper proposes distributed evolutionary digital filters (EDFs) as an improved version of the original EDF. The EDF is an adaptive digital filter which is controlled by adaptive algorithm based on evolutionary computation. In the proposed method, a large population of the original EDF is divided into smaller subpopulations. Each sub-EDF has one subpopulation and executes the small-sized main loop of the original EDF. In addition, the distributed algorithm periodically selects promising individuals from each subpopulation. Then, they migrate to different subpopulations. Numerical examples show that the distributed EDF has a higher convergence rate and smaller steady-state value of the square error than the LMS adaptive digital filter, the adaptive digital filter based on the simple genetic algorithm and the original EDF.

The capacity analysis of CDMA forward link is required to allocate high data-rate users. To solve this problem, we analyze the capacity of the CDMA forward link based on optimum power control. In a multi-cell environment, the intracell interference and intercell interference are derived. To show the capacity impact of high data-rate users, we introduce a location dependent factor, which is the ratio of intercell interference to intracell interference. Considering the location dependent factor, we propose a coordinated allocation scheme that can be used to allocate high data-rate users in sectorized cells. As a result, we show the capacity impact of high data-rate users according to location variations. The outage probability can be decreased when the coordinated allocation scheme is applied.

In this article, we survey current and next generation IX (Internet eXchange) technologies. An IX is a mechanism to interconnect many networks to each other. In other words, an ISP can establish 'peerings' with other ISPs by connecting their routers into IXes. First, we describe the basic IX model, including a policy model, called the 'bilateral' model, which allows participating ISPs to control routing policy and traffic on a 'peer' basis. Next, we classify current IX architectures from a technical point of view and discuss issues of current IXes. In the latter potion of this article, we describe next generation IX technologies, which achieve new features for IXes, such as: enabling larger volume traffic exchange with optical technology, providing virtual private peerings, migrating data-link media to participate into an IX, and exchanging traffic over widely distributed areas. We survey cutting-edge technologies for next generation IXes, and discuss the future of IX technology.

Layering architecture of the Internet Protocol provides independent processing for each layer by concealing state information from each layer. Application Program Interface (API) is based on the idea of layering model. However, the idea disturbs efficient processing of applications requiring adaptation to network environment, such as context-aware applications in mobile computing. To address this need, this paper proposes MIBsocket designed as an integrated control and management system for general network information. Any variable information related to network is shared between each applications and operating system. MIBsocket detects and informs changes in network resources to applications. MIBsocket functions such as Get, Set, and Trap used in the application adapts dynamically to any environments. MIBsocket provides portability and facility for applications: it only requires a few modification to the operating system and its API has the same programmable interfaces as usual API's. MIBsocket does not interfere the current layering architecture, but it achieves remarkable improvement on the current model. We have measured costs of MIBsocket, and applied MIBsocket to the application named interface switching system which enables a mobile host to connect to the Internet anytime by switching interface depending on network environment. As a result, the costs of MIBsocket was found acceptable. The application shows that MIBsocket is useful for movement detection and reconfiguration of network resources in the mobile computing.

Improvements of Internet technology during the last decade have shifted the technical focus from reachability to the quality of communication. There are many technical frameworks, such as Integrated Service and Differentiated Services, which have been standardized to assure the quality of communication. QoS routing is also one of such frameworks. It changes or fixes a route that IP datagrams take, and is also indispensable to put a variety of services into practice. Nevertheless, experiment reports of QoS routing on operational network are quite few, especially with dynamic SLA. Therefore, we still do not know much about the important factors for QoS-enabled network to be realized, such as users' behavior, suitable services to offer, and configuration parameters. In this paper, we carried out field-trial with pseudo QoS routing and dynamic SLA in an actual network built at the WIDE retreat in autumn 2000. In this field-trial, we provided two different types of links to attendees. Attendees chose one of the links, through which their flows go, with our dynamic SLA. We describe the details and the results of this experiment. Our results could help to understand the customers' behavior for differentiated services, and therefore be useful for designing and deploying various QoS technologies.

In wideband direct sequence code division multiple access (W-CDMA), employing an adaptive antenna array is a very promising technique to reduce severe multiple access interference (MAI) especially from high rate users. This paper proposes a fast and accurate two-step beam tracking algorithm implemented in a pilot symbol-assisted coherent adaptive antenna array diversity (CAAAD) receiver and evaluates its performance both by computer simulation and laboratory experiments. In the proposed scheme, the receiver antenna weights are updated by using both the signal-to-interference power ratio (SIR) measurements employing multiple sets of antenna weights (MSAW) and an adaptive algorithm based on the minimum mean square error (MMSE) criterion, in which other sets except for a original set of antenna weights are simply generated by a original set. Computer simulation results show that antenna weights of a four-antenna CAAAD receiver using the proposed beam tracking algorithm tracks changes in the direction of arrival (DOA) of the desired user at up to 34.3 degrees/sec, which corresponds to 215 km/h at 100 m from a base station. We also confirm based on the experiments in a radio anechoic room that the generated antenna weights track the DOA changes up to 12.3 degrees/sec.

Most of the previous controllers proposed for output regulation problems on uncertain nonlinear systems tried to keep the state variables to the nominal equilibrium points. In this letter, however, the dynamic state feedback controller makes the state variables follow the perturbed equilibrium points computed from an equilibrium-estimator.

In this paper we propose an integration of face identification and facial expression recognition. A face is modeled as a graph where the nodes represent facial feature points. This model is used for automatic face and facial feature point detection, and facial feature points tracked by applying flexible feature matching. Face identification is performed by comparing the graphs representing the input face image with individual face models. Facial expression is modeled by finding the relationship between the motion of facial feature points and expression change. Individual and average expression models are generated and then used to identify facial expressions under appropriate categories and the degree of expression changes. The expression model used for facial expression recognition is chosen by the results of face identification.

In forthcoming home network environment, computation capability will be embedded invisibly in home appliances, sensors, walls, ceilings, and floors. People will conduct various tasks using multiple devices simultaneously without consciousness of using computers. In this paper, first, we propose an application model named Virtual Network Appliance (VNA) model which simplify and expand device utilization. In the model, each device has VNA runtime system and function objects, called VNA components, running on it. A user task is defined in an application called VNA which is a logical appliance consisting of abstract function requirements and a message graph among them. Second, we propose Virtual Plug&Play mechanism which is a dynamic service integration mechanism in VNA model implementation. When a user conducts a task, he/she makes a VNA runtime system on a user-side terminal load a VNA definition appropriate for the task. Virtual Plug&Play dynamically discovers required VNA components and establishes the message graph as defined. Since XML documents are used to describe a VNA, users can share and customize it easily. We call the device integration done by Virtual Plug&Play top-down integration, which existing middleware do not aim at. Finally, we show that Virtual Plug&Play affords practical performance for top-down integration by performance evaluation.

In the current digital mobile communication that is used in the micro cellular system, a Self-Organized Dynamic Channel Assignment (DCA) Method has been proposed to use frequencies effectively. However, its characteristics and operational matters have not been reported yet. This paper takes up the TDMA/TDD system used in the current PHS system and also evaluates the characteristics and operational matters of this method through the actual operational tests. In addition, this paper aims to propose a new DCA method in order to speed up the Channel Segregation and evaluates its effects through the actual operational tests.

This paper discusses the teletraffic characteristics of cellular systems using Dynamic Channel Assignment. In general, it is difficult to exactly and theoretically analyze the teletraffic characteristics of Dynamic Channel Assignment. Also, it is not easy to theoretically evaluate influence of mobility on the traffic characteristics. This paper proposes approximate techniques to analyze teletraffic characteristics of Dynamic Channel Assignment considering mobility. The proposed techniques are based on Clique Packing approximation.

Wireless LANs have been used for realizing fully-distributed users in a multimedia environment that has the ability to provide real-time bursty traffic (such as voice or video) and data traffic. In this paper, we present a new realistic and detailed system model and a new effective analysis for the performance of wireless LANs which support multimedia communication with non-persistent carrier sense multiple access with collision avoidance (CSMA/CA) protocol. In this CSMA/CA model, a user with a packet ready to transmit initially sends some pulse signals with random intervals within a collision avoidance period before transmitting the packet to verify a clear channel. The system model consists of a finite number of users to efficiently share a common channel. Each user can be a source of both voice traffic and data traffic. The time axis is slotted, and a frame has a large number of slots and includes two parts: the collision avoidance period and the packet transmission period. A discrete-time Markov process is used to model the system operation. The number of slots in a frame can be arbitrary, dependent on the chosen lengths of the collision avoidance period and packet transmission period. Numerical results are shown in terms of channel utilization and average packet delay for different packet generation rates. They indicate that the network performance can be improved by adequate choice of ratios between the collision avoidance period and transmission period, and the pulse transmission probability.

This paper evaluates high-speed broadband packet wireless access in the forward link using coherent Time Division-Orthogonal Frequency and Code Division Multiplexing (TD-OFCDM) by applying time-multiplexed pilot symbol assisted channel estimation and integrating efficient multi-level modulation, hybrid automatic repeat request (ARQ), and code-multiplexing over a 50-100 MHz bandwidth. Computer simulation results first clarify that the common time-multiplexed pilot symbols with the transmit power of 6 dB higher than that of data symbols should be placed at both the beginning and end of a packet, and that the optimum averaging interval of channel estimates in the frequency domain is different according to the delay spread of a channel. Based on these optimized parameters for packet transmission, we show that the orthogonality among the code-multiplexed channels is destroyed due to severe frequency selective (multipath) fading and the accumulation of spread signals using equal gain combining (EGC) in the frequency domain. This degrades the achievable throughput performance especially when employing multi-level modulation and a high coding rate. Consequently, coherent TD-OFCDM with 8PSK data modulation and the convolutional coding of rate R = 2/3 employing sixteen-code multiplexing (spreading factor (SF) is 16) achieves the highest throughput of approximately 105 Mbps at the average received Eb/N0 (signal energy per bit-to-noise power spectrum density ratio) of approximately 24 dB in a 3-path Rayleigh fading channel (rms delay spread, σ= 0.1 µsec). Furthermore, in coherent TD-OFCDM with QPSK and R = 4/5 or 8PSK and R = 1/2, throughput performance greater than 80 Mbps is achieved at the average received Eb/N0 of approximately 20 dB even in a 24-path Rayleigh fading channel (σ= 0.2 µsec).

This paper proposes and investigates optimum modulation assignment and band allocation scheme according to subband channel status for BST-OFDM system. The proposed system can adaptively optimize modulation assignment and band allocation according to the conditional parameter under independently fading subband channels. Specifically, in this paper only two optimization problems are treated in terms of modulation assignment. At first, an optimization criterion is a total Bit Error Rate (BER) subject to the constraint conditions of a desired total information bit rate under a fixed effective bandwidth. Another optimization problem is the maximization of a total information bit rate to satisfy a desired BER under a fixed effective bandwidth. Knowledge of the subband channel status is assumed to be updated by the feedback information from a receiver. This paper shows that the proposed system can overperform the conventional system in which all subbands employ the same modulation schemes in terms of BER. In addition, it is shown that the proposed system improves the overall information bit rate, which is not accomplishable in the conventional system.