Application-level multicast (ALM) is a novel technology for multipoint applications, such as large scale file distribution, video and audio streaming, and video conferencing. Although many ALM mechanisms or algorithms have been proposed, all the multicast functions have been independently developed and integrated into individual applications. In such a situation, the development of ALM applications includes a lot of redundancy. Our goal is to improve the efficiency of developing ALM applications by reducing the development redundancy and to provide application developers with a middleware on which various ALM applications can be efficiently developed with minimum efforts. To this end, we develop a functional unit oriented ALM middleware, namely RelayCast. RelayCast provides a minimum but fundamental set of functionality as a functional unit, and constructs the basis on which additional and specific functions (i.e. codec, video capture, etc.) for each application are implemented. Some functional units contain several components with different algorithms, and RelayCast meets the requirements of various applications by choosing the appropriate component. In this paper, we propose RelayCast architecture, and present the implementation and experiments of a prototype.

In medium access control (MAC) protocols for wireless local area networks (WLANs), the Round-Robin scheme is the general polling scheme. A major drawback of this scheme is that it is inefficient when only a small number of stations have packets to transmit. This inefficiency is caused by polling stations that have no packets to transmit. This paper proposes an effective and simple polling scheme to reduce the number of polling times for a station with no packets to transmit. For example, the simulation result shows that the throughput increases by 35.8% when fifteen stations out of thirty stations in the polling list have packets to transmit at IEEE 802.11a 54 Mbps rate.

Incremental modification and optimization in VLSI physical design is of fundamental importance. Based on the O-tree (ordered tree) representation which has more prominent advantages in comparison with other topological representations of non-slicing floorplans, in this paper, we present an incremental placement algorithm for BBL (Building Block Layout) design in VLSI physical design. The good performance of experimental results in dealing with some instances proves the effectiveness of our algorithm.

In OFDM systems, employing a cyclic prefix (CP) as the guard interval is a simple way to combat the inter-symbol interference (ISI) and the inter-carrier interference (ICI), however it reduces the transmission efficiency of the system, especially for some channels with a very long delay spread. In this paper, we consider the OFDM system with insufficient CP, much more efficient than conventional OFDM systems. First, we present the system mathematical model and give the ISI and ICI analysis. Then the signal-to-interference power ratio (SIR) performance is presented. To reduce the ISI and ICI due to the insufficient CP, we develop a minimum-mean-square-error decision feedback equalizer (MMSE_DFE). Based on the MMSE criterion, the optimum feedforward and feedback filter coefficients are derived. For time-varying channel, to avoid brute force matrix inversion in conventional schemes, we propose an adaptive LMS based solution to update the filtering coefficients by tracing the channel variation. Since the high complexity of MMSE_DFE, a reduced complexity scheme, ordered successive partial interference cancellation DFE (OSPIC_DFE), is developed. From the performance comparison between the MMSE_DFE and the OSPIC_DFE, we see that the latter is very near to the former. Finally the simulation shows these proposed methods are highly effective in combating ISI and ICI with low complexity.

This paper introduces new methods of robust speech recognition using discrete-mixture HMMs (DMHMMs). The aim of this work is to develop robust speech recognition for adverse conditions that contain both stationary and non-stationary noise. In particular, we focus on the issue of impulsive noise, which is a major problem in practical speech recognition system. In this paper, two strategies were utilized to solve the problem. In the first strategy, adverse conditions are represented by an acoustic model. In this case, a large amount of training data and accurate acoustic models are required to present a variety of acoustic environments. This strategy is suitable for recognition in stationary or slow-varying noise conditions. The second is based on the idea that the corrupted frames are treated to reduce the adverse effect by compensation method. Since impulsive noise has a wide variety of features and its modeling is difficult, the second strategy is employed. In order to achieve those strategies, we propose two methods. Those methods are based on DMHMM framework which is one type of discrete HMM (DHMM). First, an estimation method of DMHMM parameters based on MAP is proposed aiming to improve trainability. The second is a method of compensating the observation probabilities of DMHMMs by threshold to reduce adverse effect of outlier values. Observation probabilities of impulsive noise tend to be much smaller than those of normal speech. The motivation in this approach is that flooring the observation probability reduces the adverse effect caused by impulsive noise. Experimental evaluations on Japanese LVCSR for read newspaper speech showed that the proposed method achieved the average error rate reduction of 48.5% in impulsive noise conditions. Also the experimental results in adverse conditions that contain both stationary and impulsive noises showed that the proposed method achieved the average error rate reduction of 28.1%.

There is "Processing speed improvement of the automatic fare collection gate (AFC gate)" as one of the important problems to correspond to the passengers getting on and off in high density transportation at the peak. On the other hand, reliability is indispensable to handle the ticket that is the note. Therefore, the ticket system that has both high-speed processing and high reliability is necessary and indispensable. For the passenger's convenience improvement and maintenance cost reduction, wireless IC card ticket system is hoped. However, the high-speed processing and the high reliability are ambivalent at this system because of wireless communications between an IC card and an AFC gate; the faster the AFC gate processes the ticket, the poorer the reliability gets. In this thesis, it proposes the autonomous decentralized processing technology to meet high-speed processing in wireless IC ticket system and the requirement of high reliability. "IC card" "AFC" and "Central server" are assumed to be an autonomous system. It proposes "Decentralized algorithm of the fare calculation by IC card and the AFC" to achieve high-speed processing. Moreover, "Autonomous, decentralized consistency technology" in each subsystem is shown for high-reliability. In addition, to make these the effective one, "Wireless communication area enhancing technology (touch & going method)" and "Command system for the data high speed processing" are shown. These technologies are introduced into the Suica system of East Japan Railway and the effectiveness has been proven.

High-speed intercity railways have grown into profitable business, achieving a renaissance in rail transport. High-speed railways need constant updating to new systems if they are to be winners in this age of competing transportation modes. In view of that situation, JR East started an R&D project to achieve even faster speed--more than 300 km/h. A test train that can run at an operational speed of 360 km/h is under development, and JR East plans to commence high-speed tests in the summer of 2005.

This paper proposes a logic synthesis technique for asymmetric slope differential dynamic logic (ASDDL) circuits. The technique utilizes a commercially available logic synthesis tool that has been well established for static CMOS logic design, where an intermediate library is devised for logic synthesis likely as static CMOS, and then a resulting synthesized circuit is translated automatically into ASDDL implementation at the gate-level logic schematic level as well as at the physical-layout level. A design example of an ASDDL 16-bit multiplier synthesized in a 0.18-µm CMOS technology shows an operation delay time of 1.82 nsec, which is a 32% improvement over a static CMOS design with a static logic standard-cell library that is finely tuned for energy-delay products. Design with the 16-bit multiplier led to a design time for an ASDDL based dynamic digital circuit 300 times shorter than that using a fully handcrafted design, and comparable with a static CMOS design.

Service registration and discovery are functionalities central to any service-oriented architecture, and they are often provided by centralized entities in today's systems. However, there are advantages of scalability, robustness, as well as distribution of control and cost by further decentralization of these functionalities to all the participants in the system. Peer-to-peer networks are great enablers toward this goal as they are designed to be scalable and autonomic; redundancy and automatic reconfiguarion are built into these systems, enabling peers to form and maintain the network autonomously. This article describes a fully decentralized service directory infrastructure built on top of the peer-to-peer protocol Chord. Service registration is performed implicitly by embedding semantic information into the peer identifiers, grouping peers by service categories and forming islands on the ring topology. Service discovery is performed by sending queries and anycast messages to peers registered in the appropriate islands. The routing protocol is further modified to take advantage of the island topology, with reputation mechanism and multi-path routing implemented to avoid the threat of misbehaving peers dropping transit messages in the system. Simulations were performed to assess the efficacy of both the new routing scheme and misbehavior avoidance.

The Reed-Solomon code is a versatile channel code pervasively used for communication and storage systems. The bit-serial Reed-Solomon encoder has a simple structure, although it is somewhat difficult to understand the algorithm without considerable theoretical background. Some professionals and students, not able to understand the algorithm thoroughly, might need to implement the bit-serial encoder for themselves. In this letter, a step-by-step method is presented for the implementation of the bit-serial encoder even without understanding the internal algorithm, which would be helpful for VHDL, DSP, and simulation programming.

In this paper, we are concerned with a structural ambiguity problem of tree adjoining grammars (TAGs), which is an essential problem when we try to model consensus structures of given set of ribonucleic acid (RNA) secondary structures by TAGs. RNA secondary structures can be represented as strings with structural information, and TAGs have a descriptive capability of this kind of strings, what we call structure-annotated strings. Thus, we can model RNA secondary structures by TAGs. It is sufficient to use existing alignment methods for just computing the optimal alignment between RNA secondary structures. However, when we also want to model the resulting alignment by grammars, if we adopt these existing methods, then we may fail in modeling the alignment result by grammars. Therefore, it is important to introduce a new alignment method whose alignment results can be appropriately modeled by grammars. In this paper, we will propose an alignment method based on TAG's derivations each corresponding to a given RNA secondary structure. For an RNA secondary structure, there exist a number of derivations of TAGs which correspond to the structure. From the grammatical point of view, the property of TAGs drives us to the question how we should choose a derivation from these candidates in order to obtain an optimal alignment. This is the structural ambiguity problem of TAGs, which will be mainly discussed in this paper. For dealing with this problem appropriately, we will propose an edit distance between two structure-annotated strings, and then present an algorithm which computes an optimal alignment based on the edit distance.

We propose utilizing subband-based blind source separation (BSS) for convolutive mixtures of speech. This is motivated by the drawback of frequency-domain BSS, i.e., when a long frame with a fixed long frame-shift is used to cover reverberation, the number of samples in each frequency decreases and the separation performance is degraded. In subband BSS, (1) by using a moderate number of subbands, a sufficient number of samples can be held in each subband, and (2) by using FIR filters in each subband, we can manage long reverberation. We confirm that subband BSS achieves better performance than frequency-domain BSS. Moreover, subband BSS allows us to select a separation method suited to each subband. Using this advantage, we propose efficient separation procedures that consider the frequency characteristics of room reverberation and speech signals (3) by using longer unmixing filters in low frequency bands and (4) by adopting an overlap-blockshift in BSS's batch adaptation in low frequency bands. Consequently, frequency-dependent subband processing is successfully realized with the proposed subband BSS.

Fixed polarity Reed-Muller expressions (FPRMs) exhibit several useful properties that make them suitable for many practical applications. This paper presents an exact minimization algorithm for FPRMs for incompletely specified functions. For an n-variable function with α unspecified minterms there are 2n+α distinct FPRMs, and a minimum FPRM is one with the fewest product terms. To find a minimum FPRM the algorithm requires to determine an assignment of the incompletely specified minterms. This is accomplished by using the concept of integer-valued functions in conjunction with an extended truth vector and a weight vector. The vectors help formulate the problem as an assignment of the variables of integer-valued functions, which are then efficiently manipulated by using multi-terminal binary decision diagrams for finding an assignment of the unspecified minterms. The effectiveness of the algorithm is demonstrated through experimental results for code converters, adders, and randomly generated functions.

There are several optimization techniques available for improving rendering speed of direct volume rendering. An acceleration method using the hierarchical min-max map requires little preprocessing and data storage while preserving image quality. However, this method introduces computational overhead because of unnecessary comparison and level shift between blocks. In this paper, we propose an efficient space-leaping method using optimal-sized blocks. To determine the size of blocks, our method partitions an image plane into several uniform grids and computes the minimum and the maximum depth values for each grid. We acquire optimal block sets suitable for individual rays from these values. Experimental results show that our method reduces rendering time when compared with the previous min-max octree method.

A range top-k query returns the topmost k records in the order set by a measure attribute within a specified region of multi-dimensional data. The range top-k query is a powerful tool for analysis in spatial databases and data warehouse environments. In this paper, we propose an algorithm to answer the query by selectively traversing an aggregate R-tree having MAX as the aggregate values. The algorithm can execute the query by accessing only a small part of the leaf nodes within a query region. Therefore, it shows good query performance regardless of the size of the query region. We suggest an efficient pruning technique for the priority queue, which reduces the cost of handling the priority queue, and also propose an efficient technique for leaf node organization to reduce the number of node accesses to execute the range top-k queries.

To cut down the sidelobe level of radiation pattern, a novel adaptive algorithm is proposed for electronic steering parasitic antenna. The composite objective function in this algorithm takes both directivity and sidelobe level of pattern into account, and the steepest gradient algorithm is selected to search the optimum value of reactive load. Simulations are carried out to validate the algorithm, simulated results show that the levels of sidelobe are both below -4 dB in different beamforming cases, and the front to back ratios are better than 10 dB.

Recently, Juang proposed an efficient password authenticated key agreement scheme using smart cards for the multi-server architecture. Juang's scheme was intended to provide mutual authentication and session key agreement. Herein, we show that Juang's scheme is vulnerable to a privileged insider's attack and is not easily reparable. Furthermore, it does not provide forward secrecy and the user eviction mechanism.

In this paper, a three-way divider is proposed for a partially-corporate feed in an alternating phase-fed single-layer slotted waveguide array. The divider is placed at the middle of the feed waveguide and reduces the long line effects; the frequency bandwidth is doubled. It is a kind of cross junction with one input port and three output ports; most of the power is equally divided into the right and left halves of the feed waveguide while the rest of power goes straight into the center radiating waveguide. Based upon the moment method design of the three-way divider, an inductive post is introduced for wide band power dividing control to the radiating waveguide. Reflection is below -20 dB over a wide bandwidth of 24.3-26.3 GHz, and the range of power dividing ratio ranges from 1/43 to 1/4. The amplitude and the phase from the two output ports to the feed waveguide are well balanced, and the differences are less than 0.1 dB and 5.0 degrees, respectively. The MoM analysis and the wide band design are verified experimentally in the 4 GHz band.

This paper proposes a Group Detection (GD) algorithm with Max-Log-MAP Sphere Decoder (MLM-SD) in order to reduce the complexity of signal detection in a Multiple-Input Multiple-Output Orthogonal Frequency Division Multiplexing (MIMO-OFDM) system. The proposed algorithm divides spatial streams into multiple partial spatial streams by using Minimum Mean Square Error (MMSE) detector, followed by multiple MLM-SDs with reduced number of spatial streams. Although the spatial diversity gain in the MLM-SD degrades because of the lack of the degrees of freedom exploited by the MMSE detector, its diversity gain is recovered by combining the metrics obtained by the multiple MLM-SDs. In a MIMO wireless LAN multipath fading environment, the complexity of the proposed algorithm is 10% of that of the original MLM-SD and the performance degradation in terms of SNR is slightly less than that of the original MLM-SD in 4-by-4 MIMO architecture with 64 QAM achieving 216 Mbps. It is also found that the proposed algorithm is robust against the limitation of the number of searches in sphere decoder.

This paper describes a multiband vector quantization (VQ) technique based on inner product for wideband speech coding at 16 kb/s. Our approach consists of splitting the input speech into two separate bands and then applying an independent coding scheme for each band. A code excited linear prediction (CELP) coder is used in the lower band while a transform based coding strategy is applied in the higher band. The spectral components in the higher frequency band are represented by a set of modulated lapped transform (MLT) coefficients. The higher frequency band is divided into three subbands, and the MLT coefficients construct a vector for each subband. Specifically, for the VQ of these vectors, an inner product-based distance measure is proposed as a new strategy. The proposed 16 kb/s coder with the inner-product based distortion measure achieves better performance than the 48 kb/s ITU-T G.722 in subjective quality tests.