Kurosawa, Obana, and Ogata proposed a (k,n) threshold scheme such that t cheaters can be identified, where t (k-1)/3. Their scheme is superior to previous schemes with respect to the number of participants for identifying cheaters and the size of a share. In this paper, we improve the detectability of their scheme. By using erasure decoding and the authentication code, we show that cheaters less than k/2 can be identified. Although the size of a share is larger than that of their scheme, it is independent of n.

In quantum communication theory, a realization of the optimum quantum receiver that minimizes the error probability is one of fundamental problems. A quantum receiver is described by detection operators. Therefore, it is very important to derive the optimum detection operators for a realization of the optimum quantum receiver. In general, it is difficult to derive the optimum detection operators, except for some simple cases. In addition, even if we could derive the optimum detection operators, it is not trivial what device corresponds to the operators. In this paper, we show a realization method of a quantum receiver which is described by a projection-valued measure (PVM) and apply the method to 3-ary phase-shift-keyed (3PSK) coherent-state signals.

The hardware implementation of a proposed high dimensional discrete torus knot code was successfully realized on an ASIC chip. The code has been worked on for more than a decade since then at Aichi Prefectural University and Nagoya Institutes of Technology, both in Nagoya, Japan. The hardware operation showed the ability to correct the errors about five to ten times the burst length, compared to the conventional codes, as expected from the code configuration and theory. The result in random error correction was also excellent, especially at a severely degraded error rate range of one hundredth to one tenth, and also for high grade characteristic exceeding 10-6. The operation was quite stable at the worst bit error rate and realized a high speed up to 50 Mbps, since the coder-decoder configuration consisted merely of an assemblage of parity check code and hardware circuitry with no critical loop path. The hardware architecture has a unique configuration and is suitable for large scale ASIC design. The developed code can be utilized for wider applications such as mobile computing and qualified digital communications, since the code will be expected to work well in both degraded and high grade channel situations.

The block cipher CAST-256 based on CAST-128 was a candidate algorithm for the AES round 1. In this paper we present a first result of a differential attack on CAST-256 reduced to 9 quad-rounds. One of the three round functions of CAST-256 has differential characteristics, for which a non-zero inputxor results in a zero outputxor, with high probability. This type of characteristic is the most useful for differential attack. We also show that CAST-256 has weak keys with respect to differential attack. Thus CAST-256 reduced to 9 quad-rounds can be attacked using 2123 chosen plaintexts in the case of differentially weak keys. The time complexity is about 2100 encryptions. Immunity to differential cryptanalysis of CAST-256 is not necessarily improved compared with CAST-128. Only 5 rounds of CAST-128 can be attacked using a similar differential characteristic.

This paper proposes a new satellite communication system that enables high-speed communication in a mobile environment. The system configuration combines a terrestrial mobile network and an existing satellite system, and includes a tracking antenna that was newly developed to receive 30 Mbit/s signals from commercial communication satellites. A prototype system comprising the mobile network, the satellite system and a vehicle in which the tracking antenna is installed was constructed for purposes of evaluation and demonstration. A LAN system was incorporated in the experimental vehicle by using the tracking antenna, a satellite router and a Personal Digital Cellular phone. The validity of the proposed system was verified by the tracking antenna driving tests, system UDP tests and FTP throughput tests in a mobile environment.

We propose middleware which works on widely-used commercial off-the-shelf platforms (UDP/IP, FastEthernet, and Windows NT or commercial real-time kernels) to realize real-time distributed services for plant monitoring and control systems. It is not suitable to use TCP/IP for the systems because of its unpredictable re-transmission, while, as well known, UDP/IP does not guarantee certain arrivals of packets and it is also not acceptable for the systems. With UDP/IP, packets are lost mainly because of collisions in a network and buffer overflows. To avoid these packet losses, the middleware controls scheduling of all the packets transmitted between the nodes in a distributed system and prevents excessive collisions and buffer overflows. The middleware provides a necessary set of functions for plant monitoring and control applications. The middleware on each node in a distributed system consists of library functions and run-time modules. An application program on the node is required to use these library functions according to the rules the middleware provides. In this way the middleware can manage all the traffic among the nodes in the system. Receiving requests from the application via library functions, the run-time module of the middleware schedules transmission of messages to other nodes, avoiding unexpected delivery delays or buffer overflows. The module also guarantees application-to-application quality of service (QoS), such as transmission period and delivery deadline, required by the applications. This is achieved by assigning the resources not shared by other services to each distributed service and scheduling these resources so as not to violate the assignment. Here, resources include maximum numbers of packets which a node can receive or send in a specific period (20 msec, for example). We show implementation of the middleware to make it clear how to guarantee application-to-application QoS with some application examples.

In mobile broadcasting environments, an information server periodically broadcasts a set of data items to a large mobile client population at every broadcast cycle and mobile clients retrieve the data items they need upon arrival at the broadcast channel. In such environments, the cost of data delivery is independent of the number of clients. Many applications such as auctions and stock quotes perform read-only transactions that require the clients to read consistent and current data for accurate result. Previous concurrency control mechanisms designed for such environments ensure that the clients read consistent data, but they cannot ensure latest-bound currency which allows the clients to read the latest or most up-to-date data. In this paper, we propose an efficient concurrency control mechanism that ensures latest-bound currency as well as update consistency, which is appropriate for the mobile broadcasting environments. To ensure latest-bound currency, the server computes control information using "virtual" broadcast cycles. This control information is also used for checking update consistency. Thus, the proposed mechanism allows all data committed in current broadcast cycle to be broadcast. We have performed simulation experiments to measure transaction aborts in order to evaluate the performance of the proposed mechanism. The result confirms that the proposed mechanism produces no stale reads and also shows that the proposed mechanism generates less transaction aborts than previous mechanisms, which implies that we can get higher data currency without increasing data inconsistency.

We propose a new conditional linearization based on left-right separated conditional term rewriting systems, in which the left-hand side and the right-hand side of a rewrite rule have separate variables. By developing a concept of weight decreasing joinability we first present a sufficient condition for the Church-Rosser property of left-right separated conditional term rewriting systems. Applying this result to conditional linearization, we next show sufficient conditions for the unique normal form property and the Church-Rosser property of non-duplicating (unconditional) term rewriting systems even if they are non-left-linear or overlapping.

This paper proposes a very small on-chip multimedia real-time OS for embedded system LSIs, and demonstrates its usefulness on MPEG-2 multimedia applications. The real-time OS, which has a conditional cyclic task with suspend and resume for interacting hardware (HW) / software (SW) of embedded system LSIs, implements the minimum set of task, interrupt, and semaphore managements on the basis of an analysis of embedded software requirements. It requires only about 2.5 Kbytes memory on run-time, reduces redundant conventional cyclic task execution steps to about 1/2 for HW/SW interactions, and provides sufficient performance in real-time through implementing two typical embedded softwares for practical multimedia system LSIs: an MPEG-2 system protocol LSI and an MPEG-2 video encoder LSI. This on-chip multimedia real-time OS with 2.5 Kbyte memory will be acceptable for future multimedia embedded system LSIs.

Off-line state identification methods for a sequential machine using a homing sequence or an adaptive homing sequence (AHS) are well-known in the automata theory. There are, however, so far few studies on the subject of the on-line state estimator such as a state observer (SO) which is used in the linear system theory. In this paper, we shall construct such an SO for a Moore machine based on the state identification process by means of AHSs, and discuss the convergence property of the SO.

Since the detection of optical flow (two-dimensional motion field on an image) from image sequences is essentially an ill-posed problem, most of the conventional methods use a smoothness constraint for optical flow heuristically and detect reasonable optical flow. However, little discussion exists regarding the degree of smoothness. Furthermore, to recover the relative three-dimensional motion and depth between a camera and a rigid object, in general at first, the optical flow is detected without a rigid motion constraint, and next, the motion and depth are estimated using the detected optical flow. Rigorously speaking, the optical flow should be detected with such a constraint, and consequently three-dimensional motion and depth should be determined. To solve these problems, in this paper, we apply a parametric model to an optical flow, and construct an estimation algorithm based on this model.

A new speech enhancement technique is proposed assuming that a speech signal is represented in terms of a linear probabilistic process and that a noise signal is represented in terms of a stationary random process. Since the target signal, i.e., speech, cannot be represented by a stationary random process, a Wiener filter does not yield an optimum solution to this problem regarding the minimum mean variance. Instead, a Kalman filter may provide a suitable solution in this case. In the Kalman filter, a signal is represented as a sequence of varying state vectors, and the transition is dominated by transition matrices. Our proposal is to construct the state vectors as well as the transition matrices based on time-frequency pattern of signals calculated by a wavelet transformation (WT). Computer simulations verify that the proposed technique has a high potential to suppress noise signals.

A K-band MMIC subharmonically pumped mixer integrating local oscillator (LO) amplifier has been developed. For up-converter application, it is necessary to reduce the leakage of second harmonic component of LO frequency to RF port, which is generated by nonlinear operation of LO amplifier. A quasi-lumped short-circuited stub using microstrip structure has been successfully applied to the MMIC mixer to enhance 2fLO-suppression. We propose a new configuration of a quasi-lumped short-circuited stub, which reduces the influence of parasitic elements of via-holes. The developed MMIC has a one-stage LO amplifier and it has shown about 10 dB-improvement of 2fLO-suppression compared to conventional configuration using a quarter-wavelength short-circuited stub.

Broadcast data delivery is attractive for large-size data distribution where a large user community is connected to a server through a network. It is important to consider a broadcast scheduling method which minimizes the average response time. The scheme should also guarantee the expected waiting time at the time of request. In this paper, we propose a method which divides all titles into several groups and assigns FIFO to each group. The proposed method can guarantee the waiting time for each user at his request, and is superior to FIFO (in high load) and a fixed allocation method (in low load).

To develop a smoothing method for speckle reduction is a significant problem, because of the complex ultrasonic characteristics and the obscurity of the tissue image. This paper presents a new method for speckle reduction from medical ultrasonic image by using fuzzy morphological speckle reduction algorithm (FMSR) that preserves resolvable details while removing speckle in order to cope with the ambiguous and obscure ultrasonic images. FMSR creates a cleaned image by recombining the processed residual images with a smoothed version of an original image. Performance of the proposed method has been tested on the phantom and tissue images. The results show that the method effectively reduces the speckle while preserving the resolvable details.

In this letter, we present a method to generate sets of sequences suitable for multicode transmission in quasi-synchronous (QS) CDMA systems. We focus on Gold code but extension to orthogonal Gold code is straightforward. We show that by appropriate classification of sequences, it is possible to have sets whose cross correlation is small in QS situations.

This letter presents a channel estimation and a DC-offset estimation technique in a short-ranged Bluetooth system. Each of the Bluetooth devices in the connection state knows the access codes used in the ad-hoc networks, which is utilized as a reference signal for the parameter estimation. The proposed estimators can be implemented without degradation of frame and spectral efficiency thanks to using the access code specified for the Bluetooth system.

Processes in image recognition include target detection and shape extraction. Active Net has been proposed as one of the methods for such processing. It treats the target detection in an image as an energy optimization problem. In this paper, a problem of the conventional Active Net is presented and the new Active Net is proposed. The new net is improved the ability for detecting a target. Finally, the validity of the proposed net is confirmed by experimental results.

This paper is concerned with fuzzy modeling in some reduction methods of inference rules with gradient descent. Reduction methods are presented, which have a reduction mechanism of the rule unit that is applicable in three parameters--the central value and the width of the membership function in the antecedent part, and the real number in the consequent part--which constitute the standard fuzzy system. In the present techniques, the necessary number of rules is set beforehand and the rules are sequentially deleted to the prespecified number. These methods indicate that techniques other than the reduction approach introduced previously exist. Experimental results are presented in order to show that the effectiveness differs between the proposed techniques according to the average inference error and the number of learning iterations.

This paper discusses a communication system with a multiple-access channel where two users simultaneously send complex-valued signals in the same frequency-band. In this channel, ambiguity in decoding occurs when receiver trying to estimate each users' signal. In order to solve the ambiguity problem, a family of uniquely decodable code is derived in this paper. The uniquely decodable code is designed by using trellis-coded modulation (TCM) pair where the trellis structure of one TCM is a transformation of the other in the pair. It is theoretically proved that, with the proposed coding scheme, the composite received signal can be uniquely decomposed into the two constituent signals for any power ratio and any phase difference between the received two users' signals. Improvement of BER performance over non-uniquely decodable code is illustrated by computer simulation.