In variable-length coding, the probability of codeword length per source letter being above (resp. below) a prescribed threshold is called the overflow (resp. the underflow) probability. In this paper, we show that the infimum achievable threshold given the overflow probability exponent r always coincides with the infimum achievable fixed-length coding rate given the error exponent r, without any assumptions on the source. In the case of underflow probability, we also show the similar results. From these results, we can utilize various theorems and results on the fixed-length coding established by Han for the analysis of overflow and underflow probabilities. Moreover, we generalize the above results to the case with overflow and underflow probabilities of codeword cost.

A universal channel decoder for a given family of channels is a decoder that can be designed without prior knowledge of the characteristics of the channel. Nevertheless, it still attains the same random coding error exponent as the optimal decoder tuned to the channel. This paper investigates the duality between universal channel decoders and universal source encoders. First, for the family of finite-state channels, we consider a sufficient condition for constructing universal channel decoders from universal source encoders. Next, we show the existence of a universal channel code that does not depend on the choice of the universal decoder.

A 15-50 GHz-band GaAs MMIC variable attenuator has been developed for microwave and millimeter-wave wireless communications systems. The attenuator employs a balanced distributed configuration using shunt connected HEMT's in order to realize a broadband operation, a low insertion loss and a good impedance matching with simple control bias scheme. The MMIC was fabricated with a pseudomorphic HEMT device technology. It has exhibited an insertion loss as low as 1.6 dB with an attenuation control range as wide as 21 dB at 26 GHz. It has also shown a good linearity of an input power of more than 12 dBm at 1-dB compression point and that of 26.3 dBm at a 3rd-order intercept point.

We describe a new system for high-speed wireless access systems between base stations and mobile terminals. In the proposed system, the base station has an array antenna and tracks mobile terminals by using a new tracking algorithm. A radio-on-fiber technique is used to simplify and miniaturize the components of the base station. Estimating the direction-of-arrival of the signals from a mobile terminal is important in implementing the proposed system. We propose a new tracking algorithm that uses directions-of-arrival, angular velocities of mobile terminals, and scatter modeling in multipath communications channels to improve the tracking performance. We also developed experimental equipment to demonstrate the feasibility of the proposed millimeter-wave broadband wireless access system and the efficiency of the tracking algorithm using an array antenna system. In this paper, we describe our system and present a new approach for tracking mobile terminals, which is the key feature of the system. We also discuss our simulation and experimental results.

This paper describes an electromagnetically coupled microstrip divider that provides high output port isolation and DC cutting. The device consists of a parasitic resonator placed above microstrip patch resonators, achieving tight coupling for both input and output ports. FDTD simulation and measurements reveal that the device has a high isolation between output ports. Equal and unequal 2-way and 3-way power dividers are presented in this paper.

A coded modulation scheme based on a low density parity check (LDPC) code is presented. A modified sum-product algorithm suitable for the LDPC-coded modulation scheme is also devised. Several simulation results show the excellent decoding performance of the proposed coding scheme. For example, an LDPC-coded 8PSK scheme of block length 3976 symbols achieves the symbol error probability 10-5 at only 1.2 dB away from the Shannon limit of the channel.

Although research into multimodal interfaces has been around for a long time, we believe that some basic issues have not been studied yet, e.g. the choice of modalities and their combinations is usually made without any quantitative evaluation. This study seeks to identify the best combinations of modalities through usability testing. How do users choose different interaction modes when they work on a particular application? Two experimental evaluations were conducted to compare interaction modes on a CAD system and a map system respectively. For the CAD system, the results show that, in terms of total manipulation time (drawing and modification time) and subjective preferences, the "pen + speech + mouse" combination was the best of the seven interaction modes tested. On the map system, the results show that the "pen + speech" combination mode is the best of fourteen interaction modes tested. The experiments also provide information on how users adapt to each interaction mode and the ease with which they are able to use these modes.

A method is described that can allocate bandwidth to each user flow fairly in a scalable network architecture such as differentiated services architecture. As promising queueing techniques for providing differentiated services, class-based packet scheduling and selective packet discarding have been attracting attention. However, if we consider that bandwidth should be allocated to each flow in a weighted manner, the parameters used in these methods such as the weight assigned to each class queue should be pre-determined appropriately based on an assumption about the number of flows in each class. Thus, when the actual traffic pattern differs from the assumed one, they may not work well. Instead of assuming the traffic conditions, our method estimates the number of active flows in each class by simple traffic measurement and dynamically changes the weight assigned to each class queue based on the estimated number. Our method does not need to maintain the per-flow state, which gives it scalability. Simulation showed that this method is effective under various patterns of the number of active flows.

In this paper, we propose a new type of hybrid ARQ protocol, in which a channel-adaptive variable rate channel coding scheme is combined with a multi-copy retransmission strategy so as to enhance the system performance under the delay constraint of real-time ATM services in broadband radio access networks. The main feature of a multi-copy retransmission strategy in this scheme is to improve the average throughput for a given Forward Error Correction (FEC) rate, subject to the prescribed cell loss requirement of the real-time wireless ATM services, while augmenting the reliability of channel state information required for a channel-adaptive FEC scheme. Our analysis shows that under a harsh fading channel, the proposed approach is useful for achieving the prescribed cell loss performance without significantly degrading the average throughput performance.

Simulation based education and training, especially wargame simulations, are being used widely in the field of defense modeling and in simulation communities. In order to efficiently train students and trainees, the wargame simulations must have both high performance and high fidelity. In this paper, we discuss design and implementation issues for a prototype of a parallel and distributed wargame simulation system. This wargame simulation system is based on High Level Architecture (HLA) and employs some optimization to achieve both high performance and high fidelity in the simulation system. The results show that the proposed optimization method is effective when optimization is applied to 93.5% or less of the moving objects (PFs) within the range of detection (RofD) of both the red and blue teams. Specifically, when each team has 1000 PFs we found that if the percentage of PFs within RofD is less than 50% for both teams, our method is over two times better than for the situation where there is no optimization.

Due to the advancements in Information Technology, such as the Internet and the presence of fierce competition in the market, the business environment is changing rapidly. To cope with these dynamic changes, heterogeneous systems are now required to integrate in order to form alliances with different business units or within individual business units. Since business operations can not be stopped to carry out these changes in the existing systems, the systems are required to integrate flexibly, preserving each constituent's individual characteristics. By implementing Atomic Action through a gateway and across constituent systems in a Heterogeneous Autonomous Decentralized System (HADS), higher degrees of assurance can be achieved through cooperation. However, if all the transactions are passed through a gateway, the response time worsens and the result cannot be obtained within an appropriate timeframe. Hence, a new technique of suppressing the flow passing through the gateway, while achieving a maximum number of successful transaction within the appropriate timeframe, is required.

A high assurance on-line recovery technology for a space on-board computer that can be realized using commercial devices is proposed whereby a faulty processor node confirms its normality and then recovers without affecting the other processor nodes in operation. Also, the result of an evaluation test using the breadboard model implementing this technology is reported. Because this technology enables simple and assured recovery of a faulty processor node regardless of its degree of redundancy, it can be applied to various applications, such as a launch vehicle, a satellite, and a reusable launch vehicle. As a result, decreasing the cost of an on-board computer is possible while maintaining its high reliability.

We present a 5.8 GHz VCO for the 5 GHz HIPERLAN/2 and U-NII band. The VCO uses a center-tapped inductor and a substrate shield to improve phase noise. Sensitivity to supply voltage and temperature is reduced by an amplitude control block. The design is based on a distributed inductor model which allows optimization without antecedent inductor measurements. The circuit is fabricated in a 0.8 µ m 45 GHz fT low-cost SiGe-HBT technology and operates with a supply voltage of -2.0 V to -3.3 V .

This paper proposes a vector quantization scheme which makes it possible to consider the dynamics of input vectors. In the proposed scheme, a linear transformation is applied to the consecutive input vectors and the resulting vector is quantized with a distortion measure defined by the statistics. At the decoder side, the output vector sequence is determined using the statistics associated with the transmitted indices in such a way that a likelihood is maximized. To solve the maximization problem, a computationally efficient algorithm is derived. The performance of the proposed method is evaluated in LSP parameter quantization. It is found that the LSP trajectories and the corresponding spectra change quite smoothly in the proposed method. It is also shown that the use of the proposed method results in a significant improvement of subjective quality.

In this paper we summarize the actual status of the GaN/AlGaN HFET power technology at DaimlerChrysler using high quality structures grown by MOCVD on sapphire or semi-insulating SiC supplied by Epitronics. High mobilites and record extrinsic transconductance and current values were achieved on devices with sapphire and SiC substrate. Small area devices with 0.3 µm gate length yield fT=43 GHz and fmax=78 GHz (s.i. SiC substrate). Load-pull characterisation have been performed on multifinger HFETs up to 20 GHz with e.g. output power levels above 3 Watt cw at 15 GHz for a single 1.6 mm device. Though the achieved results are very promising, the performance of these devices is still hampered by transient effects on different time scales. We will show in this paper that passivation of the devices by SiN could considerably improve the RF power performance as well as reduce long time constant effects present before passivation.

Near-optimality of subcodes of the cyclic Hamming codes is demonstrated on the binary additive channel whose noise process is the two-state homogeneous Markov chain, which is a model of bursty communication channels.

A new algorithm for the maximum a posteriori (MAP) decoding of linear block codes is presented. The proposed algorithm can be regarded as a conventional BCJR algorithm for a section trellis diagram, where branch metrics of the trellis are computed by the recursive MAP algorithm proposed by the authors. The decoding complexity of the proposed algorithm depends on the sectionalization of the trellis. A systematic way to find the optimum sectionalization which minimizes the complexity is also presented. Since the algorithm can be regarded as a generalization of both of the BCJR and the recursive MAP algorithms, the complexity of the proposed algorithm cannot be larger than those algorithms, as far as the sectionalization is chosen appropriately.

Autonomous Information Service System is a proposition made to cope with the continuously changing conditions of service provision and utilization in current information systems. The faded information field (FIF), sustained by push/pull mobile agent technology, is such a distributed architecture that brings high-assurance of the system through a balanced selective replication of the information. When the demand changes, the information environment is restructured so that the same response time to services for all unspecified users can be achieved whatever the demand volume. However, once the structure is fixed, dispatching the randomly incoming requests on the FIF is still required to guarantee the same quality of service. Our goal is to warrant the autonomous dispatching of the pull mobile agents to adjust the continuously evolving arrival distribution of the demand to the current information environment. In this paper, we explain the concepts and realization of autonomous navigation under the goal of an autonomous load balancing of the pull mobile agent volume in the FIF structure. The appropriateness of this method for FIF environments has been shown by simulation.

Before we gave a fast generalized minimum distance (GMD) decoding algorithm for one-point algebraic-geometry (AG) codes. In this paper, we propose another fast GMD decoding algorithm for these codes, where the present method includes an erasure deletion procedure while the past one uses an erasure addition procedure. Both methods find a minimal polynomial set of a given syndrome array, which is a candidate for an erasure-and-error locator polynomial set constrained with an erasure locator set of each size. Although both erasure addition and deletion GMD decoding algorithms have been established for one-dimensional algebraic codes such as RS codes, nothing but the erasure addition GMD decoding algorithm for multidimensional algebraic codes such as one-point AG codes have been given. The present erasure deletion GMD decoding algorithm is based on the Berlekamp-Massey-Sakata (BMS) algorithm from the standpoint of constrained multidimensional shift register synthesis. It is expected that both our past and present methods play a joint role in decoding for one-point AG codes up to the error correction bound.

The IETF Differentiated Services (DiffServ) framework achieves scalability by (1) aggregating traffic flows with coarse grain QoS on the data plane, and (2) allocating network resources with a bandwidth broker (BB) on the control plane. However, there are many issues that need to be addressed under such framework. First, it has been shown that the concatenation of strict priority (SP) scheduler of class-based queues (CBQ) can cause delay jitter unbounded under certain utilization, which is not acceptable to support the premium service (PS). Furthermore, it is not clear how such a DiffServ network can support traffic flows requiring the guaranteed service (GS), which is a desirable feature of the future Internet. This paper presents architecture and mechanisms to support multiple QoS under the DiffServ paradigm. On the data plane, we present a node architecture based on the virtual time reference system (VTRS). The key building block of our node architecture is the core-stateless virtual clock (CSVC) scheduling algorithm, which, in terms of providing delay guarantee, has the same expressive power as a stateful weighted fair queueing (WFQ) scheduler. With the CSVC scheduler as our building block, we design a node architecture that is capable of supporting integrated transport of the GS, the PS, the assured service (AS), and the traditional best effort (BE) service. On the control plane, we present a BB architecture to provide flexible resource allocation and QoS provisioning. Simulation results demonstrate that our architecture and mechanisms can provide scalable and flexible transport of integrated traffic of the GS, the PS, the AS, and the BE services.