In the present paper we shall examine the real-time restoration of biomedical signals under additive noises. Biomedical signals measured by instruments such as catheter manometers, ambulatory electrocardiographs and thermo-dilution sensors are susceptible to distortion and noise. Therefore, such signals must be restored to their original states. In the present study, nonstationary biomedical signals are observed and described using a mathematical model, and several restoration filters that are composed of a series of applications of this model are proposed. These filters restored band-limited approximations of the original signals in real-time. In addition, redundancy is introduced into these restoration filters in order to suppress additive noise. Finally, an optimum filter that accounts for restoration error and additive noise is proposed.

This paper reports on the development of an eyeglass-type infrared-controlled telephone communication interface for the disabled. This system is comprised of four major components: A) a headset; B) an infrared transmitting module; C) an infrared receiving/signal-processing module; and D) a main controller, the Intel-8951 microprocessor. The headset with a tongue-touch panel, a wireless earphone, and a wireless microphone. The infrared transmitting module utilizes a tongue touch panel via tongue-touch circuitry which is converted to an infrared beam and a low power laser (<0.1 mW) beam. The infrared receiving/signal-processing module, receives the infrared beam and fine tunes the unstable infrared beam into standard pulses which are used as control signals. The main controller is responsible for detecting the input signals from the infrared receiving/signal-processing module and verifying these signals with the mapping table in its memory. After the signal is verified, it is released to control the keys of the telephone interface. This design concept was mainly based on the idea that the use of an infrared remote module fastened to the eyeglasses could allow the convenient control of the dialing motion on the keys of a telephone's dialing-pad which are all modified with infrared receiving/signal-processing modules. The disabled are competent for some of work, such as a telephone operator. The increase of opportunity to do a job for the disabled would help them live independently.

In the visual secret sharing scheme proposed by Naor and Shamir, a secret image is encoded into shares, of which size is larger than that of the secret image and the shares are decoded by stacking them without performing any cryptographic computation. In this paper we propose a (k,n) visual secret sharing scheme to encode a black-and-white image into the same size shares as the secret image, where the reconstructed image of the proposed scheme is visible as well as that of the conventional scheme.

A T-user uniquely decodable (UD) code {C1,C2,,CT} over an integer set {0,1,,k} with arbitrary code length is developed for a multiple-access adder channel (MAAC). Each of the T users is equipped with two codewords, one of which is zero vector. The T-user UD code is used to identify users through the MAAC. It is shown that a T(f+g+1)-user UD code with code length f+g+1 can be arranged from two given T(f)-user and T(g)-user UD codes. This idea makes it possible to construct recursively a T-user UD code for an arbitrary code length n and a positive integer k. The T-user UD code includes the Jevticode.

The method solving Bessel's differential equation for calculating numerical values of the Bessel function Jν(x) is not usually used, but it is made clear here that the differential equation method can give very precise numerical values of Jν(x), and is very effective if we do not mind computing time. Here we improved the differential equation method by using a new transformation of Jν(x). This letter also shows a method of evaluating the errors of Jν(x) calculated by this method. The recurrence method is used for calculating the Bessel function Jν(x) numerically. The convergence of the solutions in this method, however, is not yet examined for all of the values of the complex ν and the real x. By using the differential equation method, this letter will numerically ascertain the convergence of the solutions and the precision of Jν(x) calculated by the recurrence method.

For the correctness of the minimum cut algorithm proposed in [H. Nagamochi and T. Ibaraki, Computing edge-connectivity of multigraphs and capacitated graphs, SIAM J. Discrete Mathematics, 5, 1992, pp. 54-66], several simple proofs have been presented so far. This paper gives yet another simple proof. As a byproduct, it can provide an O(m log n) time algorithm that outputs a maximum flow between the pair of vertices s and t selected by the algorithm, where n and m are the numbers of vertices and edges, respectively. This algorithm can be used to speed up the algorithm to compute DAGs,t that represents all minimum cuts separating vertices s and t in a graph G, and the algorithm to compute the cactus Γ(G) that represents all minimum cuts in G.

Systolic array implementations of modified Gaussian eliminations for the decoding of an (n, n-2t) RS code, including the Hong-Vetterli algorithm and the FIA proposed by Feng and Tzeng, are designed in this paper. These modified Gaussian eliminations are more easily understanding than the classical Berlekamp-Massey algorithm and, in addition, are efficient to decode RS codes for small e or e <

In this paper, soft decision decoding of linear block codes based on the reprocessing of several information sets is considered. These information sets are chosen according to the reliability measures of the received symbols and constructed from the most reliable information set, referred to as the most reliable basis. Each information set is then reprocessed by a multi-stage decoding algorithm until either the optimum error performance, or a desired level of error performance is achieved. General guidelines for the trade-offs between the number of information sets to be processed, the number of computations for reprocessing each information set, and the error performance to be achieved are provided. It is shown that with a proper selection of few information sets, low-complexity near-optimum soft decision decoding of relatively long block codes (64 N 128) can be achieved with significant reduction in computation complexity with respect to other known algorithms. This scheme, which generalizes the reprocessing of the most reliable basis with the ordered statistic algorithm proposed by Fossorier and Lin, is particularly efficient for codes with rate R 1/2.

A recursive maximum likelihood decoding (RMLD) algorithm is more efficient than the Viterbi algorithm. The decoding complexity of the RMLD algorithm depends on the recursive sectionalization. The recursive sectionalization which minimizes the decoding complexity is called the optimum sectionalization. In this paper, for a class of non-linear codes, called rectangular codes, it is shown that a near optimum sectionalization can be obtained with a dynamic programming approach. Furthermore, for a subclass of rectangular codes, called C-rectangular codes, it is shown that the exactly optimum sectionalization can be obtained with the same approach. Following these results, an efficient algorithm to obtain the optimum sectionalization is proposed. The optimum sectionalizations for the minimum weight subcode of some Reed-Muller codes and of a BCH code are obtained with the proposed algorithm.

We study some sufficient conditions of codeword lengths for the existence of a fix-free code. Ahlswede et al. proposed the 3/4 conjecture that Σi=1n a-li 3/4 implies the existence of a fix-free code with lengths li when a=2 i. e. the alphabet is binary. We propose a more general conjecture, and prove that the upper bound of our conjecture is not greater than 3/4 for any finite alphabet. Moreover, we show that for any a2 our conjecture is true if codeword lengths l1,l2,. . . consist of only two kinds of lengths.

The initial cell search procedure of a terminal in an asynchronous wideband CDMA (WCDMA) system is discussed. The procedure consists of the following steps (not necessarily in this order): chip and frame synchronization; identification and synchronization of the long scrambling code; and determination of the target base station identity. Higuchi et al. proposed a cell search method for such a system. We propose a modification of that scheme which offers substantial terminal complexity reductions with the same performance. The price is a slight increase in delay. Furthermore, we study the impact on performance and complexity for different parameter settings for these methods.

This paper proposes a fully balanced circuit structure with a zero common-mode gain. The common-mode gain of the proposed structure becomes theoretically zero with a perfect device matching. Even if a perfect device matching is not achieved, the common-mode signal can be sufficiently suppressed by the feedback loops provided with the structure. Based on this concept, an integrator is composed. Furthermore the concept can be directly applied to a filter design. The application results in reduced chip area. A design example of a second-order filter and simulation results verify the theoretical expectation.

This paper presents unified VLSI architectures which can efficiently realize some widespread one-dimensional (1-D) and two-dimensional (2-D) real discrete trigonometric transforms, including the discrete Hartley transform (DHT), discrete sine transform (DST), and discrete cosine transform (DCT). First, succinct and unrestrictive Clenshaw's recurrence formula along with the inherent symmetry of the trigonometric functions are adequately employed to render efficient recurrences for computing these 1-D RDTT. By utilizing an appropriate row-column decomposition approach, the same set of recurrences can also be used to compute both of the row transform and column transform of the 2-D RDTT. Array architectures, basing on the developed recurrences, are then introduced to implement these 1-D and 2-D RDTT. Both architectures provide substantial hardware savings as compared with previous works. In addition, they are not only applicable to the 1-D and 2-D RDTT of arbitrary size, but they can also be easily adapted to compute all aforementioned RDTT with only minor modifications. A complete set of input/output (I/O) buffers along with a bidirectional circular shift matrix are addressed as well to enable the architectures to operate in a fully-pipelined manner and to rectify the transformed results in a natural order. Moreover, the resulting architectures are both highly regular, modular, and locally-connected, thus being amenable to VLSI implementations.

In this paper, we consider sufficient conditions for ruling out some useless iteration steps in a class of soft-decision iterative decoding algorithms for binary block codes used over the AWGN channel using BPSK signaling. Sufficient conditions for ruling out the next single decoding step, called ruling-out conditions and those for ruling out all the subsequent iteration steps, called early termination conditions, are formulated in a unified way without degradation of error performance. These conditions are shown to be a type of integer programming problems. Several techniques for reducing such an integer programming problem to a set of subprograms with smaller computational complexities are presented. As an example, an early termination condition for Chase-type decoding algorithm is presented. Simulation results for the (64, 42, 8) Reed-Muller code and (64, 45, 8) extended BCH code show that the early termination condition combined with a ruling-out condition proposed previously is considerably effective in reducing the number of test error patterns, especially as the total number of test error patterns concerned grows.

Recently, it is widely known that the partial response maximum-likelihood (PRML) system has attracted much attention as one of indispensable signal processing technique for achieving high density digital magnetic recording. But, the performance of PRML system is degraded by many undesirable causes in recording channel. For improving the performance, it is desirable to use any high order PRML system or high rate code. Our proposed two-track recording method increases the coding rate over 1, and contributes to decrease these degradation effects. The recording code in our system adopts a turbo code which provides a substantial near-ML performance by the suboptimum iterative decoding algorithm. In this paper, the turbo coded class four partial response (PR4) systems using the rate 4/6, 8/10 and 16/18 codes for high density two track digital magnetic recording are proposed. The error rate performance of the proposed system is obtained by computer simulation taking account of the partial erasure, which is one of nonlinear distortions at high densities. The performance of our system is compared with that of the conventional NRZ coded PR4ML system. The result shows that the proposed system is hardly affected by partial erasure and keeps good performance in high density recording. In especial, the proposed system using the rate 16/18 turbo code can achieve a bit error rate of 10-4 with SNR of approximately 12.2 dB less than the conventional NRZ coded PR4ML systems at a normalized linear density of 3.

The subliminal channel is one of the methods for hiding a message in other messages. Simmons has shown conjectures on the upper bound to the bandwidth of a subliminal channel. This paper proposes a new broad-band subliminal channel embedded in the ESIGN. The bandwidth of the proposed subliminal channel is wider than that of the previous one, and it exceeds the upper bound that Simmons has conjectured. Namely, we disprove the conjectures due to Simmons. We also show that it is possible to construct the subliminal channel even if the transmitter and the subliminal receiver do not have any key in common.

In order to guarantee pairwise independence of codewords in an ensemble of convolutional codes it is necessary to consider time-varying codes. However, Seguin has shown that the pairwise independence property is not strictly necessary when applying the random coding argument and on this basis he derives a new random coding bound for rate 1/n fixed convolutional codes. In this paper we show that a similar random coding bound can be obtained for rate k/n fixed convolutional codes.

This paper presents an improved method of speaker verification using the sequential probability ratio test (SPRT), which can treat the correlation between successive feature vectors. The hidden Markov model with the mean field approximation enables us to consider the correlation in the SPRT, i. e. , using the mean field of previous state, probability computation can be carried out as if input samples were independent each other.

We propose a traffic measurement system which uses trap and polling methods. To obtain its performance we consider a queueing model with a single server and evaluate a packet delay. In our multi-cast traffic, packets are modeled as a batch with a batch size distribution {gk}. The batch arrival process is observed as two processes on the basis of batch size. For a batch whose size is more than or equal to a threshold L, the batch will be trapped by our traffic measurement system (in queueing model, it will enter a queue immediately after its arrival). For a batch whose size is less than L, it will be observed at a polling cycle T (in queueing model, it will be temporarily stored in a buffer and all these small batches will be cyclically noticed with a cycle T). We analyze this queueing model by a diffusion approximation and compare the packet delay observed by our traffic measurement system with the L=1 original batch arrival model. Evaluating the results of the diffusion approximation, we illustrate that our traffic measurement system has functions not only to give an accurate estimation of the mean waiting time but also reduce the number of measurements by choosing appropriate parameters L and T.

In this paper, we study the link capacity assignment problem in packet-switched networks (CA problem) focusing on the case where link cost function is a piecewise linear concave function. This type of cost function arises in many communication network design problems such as those arising from developments in communication transmission technologies. It is already known that the method of link set assignment is applicable for solving the CA problem with piecewise linear convex cost function. That is, each link in the network is assigned to one of a group of specific sets, and checked for link set contradiction. By extending the method of link set assignment to the case of piecewise linear concave cost function, an important characteristic of the optimal solution of the CA problem is derived. Based on this characteristic, the non-differentiable link cost function can be treated as a differentiable function, and a heuristic algorithm derived from the Lagrange multiplier method is then proposed. Although it is difficult to determine the global optimum of the CA problem due to its non-convexity, it is shown by numerical results that the solution obtained from the proposed algorithm is very close to the global optimum. Moreover, the computation time is linearly dependent on the number of links in the problem. These performances show that the proposed algorithm is very efficient in solving the CA problem, even in the case of large-scale networks.