The present paper investigates a two-dimensional (2-D) adaptive lattice filter used for modeling 2-D AR fields. The 2-D least mean square (LMS) lattice algorithm is used to update the filter coefficients. The proposed adaptive lattice filter can represent a wider class of 2-D AR fields than previous ones. Furthremore, its structure is also shown to possess orthogonality in the backward prediction error fields. These result in superior convergence and tracking properties to the adaptive transversal filter and other adaptive 2-D lattice models. Then, the convergence property of the proposed adaptive LMS lattice algorithm is discussed. The effectiveness of the proposed model is evaluated for parameter identification through computer simulation.

This paper presets a fast computation algorithm for connection admission control for heterogeneous delay sensitive traffic in ATM networks. Cell loss probability is adopted as the measure of quality of service. In our study, cells of each connection are allowed to have two different loss priorities and the aggregate traffic can have more than two. To cope with multiple quality of service requirements, the link capacity is divided into several bands. For simplicity, each traffic source is assumed to alternate between active and idle periods. However, the results can be extended to traffic sources having more than two states. Upper bounds of actual cell loss probabilities are derived based on the bufferless fluidflow model. Numerical results show that the upper bounds are close to the actual cell loss probabilities.

This paper presents a binary neural network approach for link activation problems in multihop radio networks. The goal of the NP-complete problems is to find a conflict-free link activation schedule with the minimum number of time slots for specified communication requirements. The neural network is composed of NM binary neurons for scheduling N links in M time slots. The energy functions and the motion equations are newly defined with heuristic methods. The simulation results through 14 instances with up to 419 links show that the neural network not only surpasses the best existing neural network in terms of the convergence rate and the computation time, but also can solve large scale instances within a constant number of iteration steps.

It is an important problem in fields of radar, sonar, and so on to estimate parameters of closely spaced multiple signals. The MUSIC algorithm with the forward-backward (FB) spatial smoothing is considered as the most effective technique at present for the problem with coherent signals in a variety of fields. We have applied this in Laser Microvision. Recently, Shimotahira has proposed the Kernel MUSIC algorithm, which is applicable to cases when signal vectors and noise vectors are orthogonal. It also utilizes Gaussian elimination of the covariance matrix instead of eigenvalue analysis to estimate noise vectors. Although the amount of computation by the Kernel MUSIC algorithm became lighter than that of the conventional MUSIC algorithm, the covariance matrix was formed to estimate noise vectors and also all noise vectors were used to analyze the MUSIC eigenspectrum. The heaviest amount of computation in the Kernel MUSIC algorithm exists in the transformation of the covariance matrix and the analysis of the MUSIC eigenspectrum. We propose a more straightforward algorithm to estimate noise vectors without forming a covariance matrix, easier algorithm to analyze the MUSIC eigenspectrum. The superior characteristics will be demonstrated by results of numerical simulation.

Efficient parallel algorithms for several problems on proper circular arc graphs are presented in this paper. These problems include finding a maximum matching, partitioning into a minimum number of induced subgraphs each of which has a Hamiltonian cycle (path), partitioning into induced subgraphs each of which has a Hamiltonian cycle (path) with at least k vertices for a given k, and adding a minimum number of edges to make the graph contain a Hamiltonian cycle (path). It is shown here that the above problems can all be solved in logarithmic time with a linear number of EREW PRAM processors, or in constant time with a linear number of BSR processors. A more important part of this work is perhaps the extension of basic BSR to allow simultaneous multiple BROADCAST instructions.

We have proposed a concept of low power drive system for a multi-media TFT-LCD using MFD in which a displayed image is divided into some interlaced subfield images and the number of interlaced subfields can be changed depending on the moving quantities of displayed images. This method has been applied to a 9.5" TFT-LCD and successful operation has been confirmed without moving image degradation.

Various reconfiguration schemes against faults of mesh-connected processor arrays have been proposed. As one of them, the mesh-connected processor arrays model based on single-track switches was proposed in [1]. The model has an advantage of its inherent simplicity of the routing hardware. Furthermore, the 2 track switch model [2] and the multiple track switch model [3] were proposed to enhance yields and reliabilities of arrays. However, in these models, Simplicity of the routing hardware is somewhat lost because multiple tracks are used for each row and column. In this paper, we present a builtin self-reconstruction approach for mesh-connected processor arrays which are partitioned into sub-arrays each using single-track switches. Spare PEs which are located on the boundaries of the sub-arrays compensate faulty PEs in these sub-arrays. First, we formulate a reconfigulation algorithm for partitioned mesh-arrays using a Hopfield-type neural network, and then its performance for reconfigulation in terms of survival rates and reliabilities of arrays and processing time are investigated by computer simulations. From the results, we can see that high reliabilites are achieved while processing time is a little and hardware overhead (links and switches) required for reconstruction is as same as that for the track switch model. Next, we present a hardware implementation of the neural algorithm so that a built-in self-reconfigurable scheme may be realized.

This paper considers the estimation of speech parameters and their enhancement using an approach based on the estimation-maximization (EM) algorithm, when only noisy speech data is available. The distribution of the excitation source for the speech signal is assumed as a mixture of two Gaussian probability distribution functions with differing variances. This mixture assumption is experimentally valid for removing the residual excitation signal. The assumption also is found to be effective in enhancing noise-corrupted speech. We adaptively estimate the speech parameters and analyze the characteristics of its excitation source in a sequential manner. In the maximum likelihood estimation scheme we utilize the EM algorithm, and employ a detection and an estimation step for the parameters. For speech enhancement we use Kalman filtering for the parameters obtained from the above estimation procedure. The estimation and maximization procedures are closely coupled. Simulation results using synthetic and real speech vindicate the improved performance of our algorithm in noisy situations, with an increase of about 3 dB in terms of output SNR compared to conventional Gaussian assumption. The proposed algorithm also may be noteworthy in that it needs no voiced/unvoiced decision logic, due to the use of the residual approach.

Because the match phase in OPS5-type production systems requires most of the system's execution time and memory accesses, we proposed hash-based parallel production systems, CPPS (Clustered Parallel Production Systems), based on the RETE algorithm for distributed memory parallel computers, or multicomputers to reduce such a bottleneck. CPPS was effective in speeding up the match phase, but still left room for optimizations. In this paper, we introduce software cache techniques to memory nodes in the CPPS as one of the optimizations, and implement it on a multicomputer, nCUBE2. The benchmark results show that the CPPS with the software cache is about 2-fold faster than the original, and more than 7-fold faster than the simple hash method proposed by Acharya et al. for a large scale problem. The speed-up can be attributed to decreased communication costs.

An attenuation correction method was proposed for laser-produced plasma emission computed tomography (ECT), which is based on a relation of the attenuation coefficient and the emission coefficient in plasma. Simulation results show that the reconstructed images are dramatically improved in comparison to the reconstructions without attenuation correction.

A classs of type 1 linear phase FIR digital filters is proposed. The filter can be realized using a parallel, modular and regular array structure. It is shown that, under some simple constraints, the consisting modules of the array can be realized free of multiplier coefficients. Such two dimensional mesh arrays are specially suitable for realization with special-purpose systolic hardware for high-speed digital signal processing tasks. Compared to the array structure, proposed by the authors, for multiplierless realization of maximally flat FIR digital filters, this class needs less adders to fulfill the same magnitude response requirements. Another attractive property of the proposed array is that a number of highpass or lowpass filters with different passband widths can be realized simultaneously in a very economical way.

This paper presents the system issues involved with the transmission of pre-encoded VBR MPEG video using CBR service. Conventional wisdom suggests that lossless delivery of VBR video using CBR service requires bandwidth to be reserved at the peak rate resulting in low bandwidth utilization. We calculate the minimum rate at which bandwidth must be reserved on a network in order to provide continuous playback of an MPEG encoded video bitstream. Simulation results using the frame size traces from several pre-encoded MPEG bitstreams and several buffer sizes demonstrate that this minimum reservation rateis much lower than the peak rate when a relatively small playback buffer size is used, resulting in much higher bandwidth utilization. Procedures for performing connection setup and lossless realtime video playback between the video server and the client are outlined. Methods for incorporating VCR-like features such as pauseandfast forward/reversefor Video-on-Demand (VoD) applications are presented.

Many activities are being promoted for the coming multimedia age. In this paper, background information for multimedia communications is followed by an outline of joint tests in multimedia communications with some examples of the projects and applications. These trials are also explained from the aspects of project specifications, which include application classifications and details of multimedia-on-demand offerings, as well as technical issues in experimental environments which mainly include those related to ATM technology.

A laser power optimizing method for multi-pulse recording is described. Multi-pulse recording uses the recording pulse formed by bias part and comb part. To obtain best readout signal characteristics and reduce the time for optimizing, new mark pattern is recorded and then two parts of the recording pulse are individually adjusted by evaluating the detected signals during pre-write testing. At the optimized laser power by this method, a good qualitative eyepattern was obtained. As a result, this new method proves to be suitable for the multi-pulse recording and adapted to various disks with different recording properties.

In this paper, DSPs, of which memory addresses are pointed by special purpose registers (address registers: ARs), are assumed, and methods to derive an efficient memory access pattern for those DSPs proposed. In such DSPs, programmers must take care for efficient allocation of memory space as well as effective use of registers, in order to derive an efficient program in the sense of execution period. In this paper, memory addresses and AR update operations are modeled by an access graph, and a novel memory allocation method is presented. This method removes cycles and forks in a given access graph, and decides an address location of variables in memory space with less overhead. In order to utileze multiple ARs, methods to assign variables into ARs are investigated. The proposed methods are applied to the compiler for DSP56000 and are proved to be effective by generated codes for several examples.

This paper proposes new algorithms for diagnosing (detection, identification and location) baseline multistage interconnection networks (MIN) as one of the basic units in a massively parallel system. This is accomplished in the presence of single and multiple faults under a new fault model. This model referred to as the geometric fault model, considers defective crossing connections which are located between adjacent stages, internally to the MIN (therefore, a fault corresponds to a physical bridge fault between two connections). It is shown that this type of fault affects the correct geometry of the network, thus requiring a different testing approach than previous methods. Initially, an algorithm which detects the presence of bridge faults (both in the single and multiple fault cases), is presented. For a single bridge fault, the proposed algorithm locates the fault except in an unique pathological case under which it is logically impossible to differentiate between two equivalent locations of the fault (however, the switching element affected by this fault is uniquely located). The proposed algorithm requires log2 N test vectors to diagnose the MIN as fault free (where N is the number of input lines to the MIN). For fully diagnosing a single bridge fault, this algorithm requires at most 2 log2 N tests and terminates when multiple bridge faults are detected. Subsequently, an algorithm which locates all bridge faults is given. The number of required test vectors is O(N). Fault location of each bridge fault is accomplished in terms of the two lines in the bridge and the numbers of the stages between which it occurs. Illustrative examples are given.

The objective of this paper is to find a parametric representation for the vocal-tract log-area function that is directly and simply related to basic acoustic characteristics of the human vocal-tract. The importance of this representation is associated with the solution of the articulatory-to-acoustic inverse problem, where a simple mapping from the articulatory space onto the acoustic space can be very useful. The method is as follows: Firstly, given a corpus of log-area functions, a parametric model is derived following a factor analysis technique. After that, the articulatory space, defined by the parametric model, is filled with approximately uniformly distributed points, and the corresponding first three formant frequencies are calculated. These formants define an acoustic space onto which the articulatory space maps. In the next step, an independent component analysis technique is used to determine acoustic and articulatory coordinate systems whose components are as independent as possible. Finally, using singular value decomposition, acoustic and articulatory coordinate systems are rotated so that each of the first three components of the articulatory space has major influence on one, and only one, component of the acoustic space. An example showing how the proposed model can be applied to the solution of the articulatory-to-acoustic inverse problem is given at the end of the paper.

This paper introduces a new method to recover 3-D road plane from its 2-D monocular perspective image. The research is aimed at the reconstruction of depth information from the 2-D visual input in road following and navigation. Planar road model is considered and the road-centered coordinate system which forms slope and turn angles with camera-centered coordinate system is used to describe boundary points on road plane. We develop approaches to find matching points of boundaries of road and to obtain angular parameters thereafter. A way of finding depth of matching points from the perspective images and angular parameters together is proposed. Therefore the 3-D road reconstruction can be replicated without introducing any parameters of inverse perspective.

The transform coding scheme is often used for data compression of images, but the blocking effects peculiar to the scheme appear more clearly in reproduced images as a coding rate (bits/pixel) decreases. These effects can sometimes be viewed as a periodical square-grid overlaying the images. In this paper,we propose a new method for selectively measuring the above blocking effects among several types of image degradation by means of the techniques of nonlinear signal processing for spectral infomation (cepstral techniques), in order to compare the amount of blocking effects for the different coding images. First a two-component model which consists of DC and AC images, is discussed from a viewpoint of subimage-by-subimage coding, and some basic properties of cepstral information for the model are investigated. Then we show a procedure to compute the cepstral information for two-dimensional image signals taking the horizontal and vertical directions ioto account, and introduce a cepstral mean square error (CMSE) as a new measure to estimate the amount of blocking effects. The computer simulation results for some test images using different coding schemes show that the amount of blocking effects in each image can be easily measured and estimated by this method even when the blocking effects appear slightly.

In this paper, quantization method which can keep the phase and gain characteristics of a reference filter is proposed. The proposed method uses a genetic algorithm and a simulated annealing algorithm. The objective function used in this method is described with two kinds of weighting functions for identifying the phase and gain characteristics respectively. Therefore, the quantization accuracy on the gain characteristic is independent of the accuracy on the phase characteristic. Further, the proposed algorithm can be applied to any types of filters, because the chromosome expresses only their coefficients values. The efficiency of the proposed algorithm is verified by some experiments.