One of the disadvantages of compressed data is their vulnerability, that is, even a single corrupted bit in compressed data may destroy the decompressed data completely. Therefore, Variable-to-Fixed length Arithmetic Coding, or VFAC, with error detecting capability is discussed. However, implementable error recovery method for compressed data has never been proposed. This paper proposes Burst Error Recovery Variable-to-Fixed length Arithmetic Coding, or BERVFAC, as well as Error Detecting Variable-to-Fixed length Arithmetic Coding, or EDVFAC. Both VFAC schemes achieve VF coding by inserting the internal states of the decompressor into compressed data. The internal states consist of width and offset of the sub-interval corresponding to the decompressed symbol and are also used for error detection. Convolutional operations are applied to encoding and decoding in order to propagate errors and improve error control capability. The proposed EDVFAC and BERVFAC are evaluated by theoretical analysis and computer simulations. The simulation results show that more than 99.99% of errors can be detected by EDVFAC. For BERVFAC, over 99.95% of l-burst errors can be corrected for l 32 and greater than 99.99% of other errors can be detected. The simulation results also show that the time-overhead necessary to decode the BERVFAC is about 12% when 10% of the received words are erroneous.

In order that speech recognition system may have a high recognition rate in a noisy environment, a wide-band sharp directional microphone system is required at the input for securing a high S/N ratio. The authors have already reported the realization of a wide-band uni-directional microphone system by three-microphone integration method. In this paper, we intend to describe the derivation of a sharp directivity function and the realization of its microphone system. First, setting the shape of the characteristic function to bring a sharp directional pattern and then expanding it into the Fourier series, we derive a new directivity function. Next, on the basis of this directivity function, we will present a sharp directional microphone system with only three non-directional microphones and the subsequent analog signal processing. And also, the directional pattern acquired by the proposed method and the effect of the dispersion in the sensitivity of the constituent microphones on the directivity are discussed in detail.

Call admission control (CAC) is becoming vital for multimedia services in the ability of wireless/mobile networks to guarantee Quality of Service (QoS) partially due to the network's limited capacity. In this paper, we propose an optimal call admission control scheme with bandwidth reallocation algorithm (multi-class-CAC-BRA) for multi-classes of adaptive multimedia services in wireless/mobile networks. The multi-class-CAC-BRA approach optimizes revenue for service providers and satisfies QoS requirements for service users. The proposed approach adopts semi-Markov Decision Process to model both call admission control and bandwidth reallocation algorithm. In other words, whenever decisions are made, decisions are made for both call admission control and bandwidth reallocation. Since the non-adaptive multimedia traffic is a special case of the adaptive multimedia traffic, the non-adaptive optimal CAC scheme is a special case of our optimal multi-class-CAC-BRA scheme. Furthermore, the Interior-point Method in linear programming is used to solve the optimal decision problem. Simulation results reveal that the proposed multi-class-CAC-BRA scheme adapts itself well to adaptive multi-class multimedia traffic, achieves optimal revenue, and satisfies QoS requirements that are the upper bounds of handoff dropping probabilities. Our approach solves the optimal adaptive multimedia CAC problem. We believe that this work has both theoretical and practical significance.

This paper proposes an adaptive zone selection (AZS) scheme with adaptive modulation for wireless packet transmission systems to achieve high throughput and low delay performances even under non-uniform traffic conditions. In the proposed system, based on the measurement of the propagation path characteristics to each access point (AP) as well as broadcasted blocking probability information from each AP, a terminal autonomously selects an AP and modulation parameters that give the minimum transmission failure probability determined by both the call blocking rate due to lack of radio resource and packet error rate due to severe channel conditions. Computer simulation confirms that the proposed scheme greatly improves throughput and delay performances especially under non-uniform traffic conditions.

The problem of finding a minimum-cast multicast tree (Steiner tree) is known as NP complete. Heuristic based algorithms for this problem to achieve good performance are usually time-consuming. In this paper, we propose a new strategy called tree-caching for efficient multicast connection setup in connection-oriented networks. In this scheme, the tree topologies that have been computed are cached in a database of the source nodes. This can reduce the connection establishment time for subsequent connection requests which have some common multicast members, by an efficient reuse of cached trees without having to re-run a multicast routing algorithm for the whole group. This method can provide an efficient way to eliminate, when ever possible, the expensive tree computation algorithm that has to be performed in setting up a multicast connection. We first formulate the problem of tree-caching and then propose a tree-caching algorithm to reduce the complexity of the tree computations when a new connection is to be established. Through simulations, we find that the proposed tree-caching strategy performs very well and can significantly reduce the computation complexity for setting up multicast connections.

This paper presents a simple and cost-effective bidirectional antenna using a probe excited circular ring. The structure of the antenna is simple i.e., a linear electric probe surrounded by the circular ring. The principle of the antenna design is easy and straightforward. A choice of the ring radius is first chosen to achieve the condition that only the dominant mode can be propagated. Furthermore, it is found that for a specific ring radius, the radiation patterns of the antenna are varied as the ring width. Then, the optimum ring width that provides the maximum directivity is determined. The criterion of the selection of the ring width for various ring radii is illustrated as the guidelines for the antenna design. The fabricated antennas at the operating frequency of 1.9065 GHz are measured and compared with the theoretical predictions. It is apparent that these results are in reasonable agreement. The bidirectional pattern with the gain of 5.4 dBi over the bandwidth of 17% is obtained. Moreover, the antenna can be easily fabricated with the low production cost. Therefore, this antenna is suitable for installing at the base station in the street cell.

This paper proposes a unified coding algorithm for the lossless and near-lossless compression of still color images. The proposed unified color image coding scheme can control the Peak Signal-to-Noise Ratio (PSNR) of the reconstructed image while the level of distortion on the RGB plane is suppressed to within a preset magnitude. In order to control the PSNR, the distortion level is adaptively changed at each pixel. An adaptive quantizer to control the distortion is designed on the basis of psychovisual criteria. Finally, experiments on Super High Definition (SHD) images show the effectiveness of the proposed algorithm.

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.

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.

We have developed a signal processing method that is appropriate for detecting electromagnetic radiation due to earthquake activities. The radiation is usually accompanied by a background noise that is mainly caused by atmospheric discharges in the tropical regions. Data representing the seismic radiation is presented as sound via the concept of sonification. This is useful for immediately finding out anomalous seismic radiations, which are often followed by a disastrous earthquake, from the massive data collected from over forty observation stations. It is illustrated that the auditory display is valuable for future earthquake prediction systems.

We have proposed and implemented a spread spectrum (SS) wireless switch using 2.4 GHz front-end AlN/Al2O3 surface acoustic wave (SAW) matched filter (MF). Since the SAW MF has radio frequency (RF) front-end operation, RF components are not needed in the received circuit. High impedance in the peripheral circuit using passive devices has been employed for low current consumption. The SS wireless switches have been designed with the power consumption of less than 100 µW by using the SAW MF. It is confirmed that implemented SS wireless switch has a long battery life of 10 years and communication range of 30 m.

The normalized least mean square (NLMS) algorithm has the drawback that the convergence speed of adaptive filter coefficients decreases when the reference signal has high auto-correlation. A technique to improve the convergence speed is to apply the decorrelated reference signal to the calculation of the gradient defined in the NLMS algorithm. So far, only the effect of the improvement is experimentally examined. The convergence property of the adaptive algorithm to which the technique is applied is not analized yet enough. This paper first defines a cost function properly representing the criterion to estimate the coefficients of adaptive filter. The name given in this paper to the adaptive algorithm exploiting the decorrelated reference signal, 'normalized least mean EE' algorithm, exactly expresses the criterion. This adaptive algorithm estimates the coefficients so as to minimize the product of E and E' that are the differences between the responses of the unknown system and the adaptive filter to the original and the decorrelated reference signals, respectively. By using the cost function, this paper second specifies the convergence condition of the normalized least mean EE' algorithm and finally presents computer simulations, which are calculated using real speech signal, to demonstrate the validity of the convergence condition.

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 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.

In this paper, important methods for cryptanalysis of the stream cipher based on a class of keystream generators are discussed. These methods employ an approach called the fast correlation attack. This cryptographic problem is treated by considering its equivalent channel coding approach, namely decoding of certain very low rate codes in presence of very high noise. A novel family of algorithms for the fast correlation attack is presented. The algorithms are based on the iterative decoding principle in conjunction with a novel method for constructing the parity-checks. A goal of this paper is to summarize reported results and to compare some of the recent ones. Accordingly, the family is compared with recently proposed improved fast correlation attacks based on iterative decoding methods. An analysis of the algorithms performances and complexities is presented. The corresponding trade-offs between performance, complexity and required inputs are pointed out.

This paper presents a Mix-net that has the following properties; (1) it efficiently handles long plaintexts that exceed the modulus size of the underlying public-key encryption scheme as well as very short ones (length-flexibility), (2) input ciphertext length is not impacted by the number of mix-servers (length-invariance), and (3) its security in terms of anonymity can be proven in a formal way (probable security). If desired, one can add robustness so that it outputs correct results in the presence of corrupt users and servers. The security is proven in such a sense that breaking the anonymity of our Mix-net is equivalent to breaking the indistinguishability assumption of the underlying symmetric encryption scheme or the Decision Diffie-Hellman assumption.

The maximum amounts of embedded information that is important in practical system design of two watermarking methods based on the parallel combinatorial spread spectrum (PC/SS) scheme are discussed in this paper. One is a private watermarking method proposed in this paper and has a practical strong point to make the quality of the watermarked image to be constant in any images. The other is a public watermarking method that was previously proposed by the authors. Through this study, the minimum number of orthogonal sequences for embedding the required amount of information under the condition that quantization noise is only assumed is found in each watermarking method.

A combination of a software and a systolic hardware implementation for the Quasi Arithmetic compression algorithm is presented. The hardware is implemented as a pipeline hardware implementation. The implementation doesn't change the the algorithm. It just split it into two parts. The combination of parallel software and pipeline hardware can give very fast compression without decline of the compression efficiency.

This paper describes a new method for estimating the direction of arrival (DOA) using a nonlinear microphone array system based on complementary beamforming. Complementary beamforming is based on two types of beamformers designed to obtain complementary directivity patterns with respect to each other. In this system, since the resultant directivity pattern is proportional to the product of these directivity patterns, the proposed method can be used to estimate DOAs of 2(K-1) sound sources with K-element microphone array. First, DOA-estimation experiments are performed using both computer simulation and actual devices in real acoustic environments. The results clarify that DOA estimation for two sound sources can be accomplished by the proposed method with two microphones. Also, by comparing the resolutions of DOA estimation by the proposed method and by the conventional minimum variance method, we can show that the performance of the proposed method is superior to that of the minimum variance method under all reverberant conditions.

In this paper, we propose an efficient channel estimation technique for orthogonal frequency-division multiplexing (OFDM) systems with transmitter diversity. The proposed technique estimates uniquely all channel frequency responses needed in space-time coded OFDM systems using "comb-type" training symbols. The computational complexity of the proposed technique is reduced dramatically, compared with the previous minimum mean-squared error (MMSE) technique, due to the processing made all in the frequency-domain. Also, several other techniques for mitigating random noise effect and tracking channel variation are discussed to further improve the performance of the proposed approach. The performances of the proposed approach are demonstrated by computer simulation for mobile wireless channels.