A single chip motion estimator is described dedicatedly for MPEG2 MP＠HL moving pictures. Adopting a two-level hierarchical searching algorithm in detecting motion vectors, the computational labor can be reduced by 1/70 in comparison with the conventional algorithm. A novel mechanism is introduced into the full-search procedure, which attempts the maximum possible reuse of reference pixels in order to reduce the bandwidth of the frame memory interface. The proposed motion estimator is integrated in a 0.6 µm triple-metal CMOS chip, which contains 1,450 K transistors on a 12.713.7 mm2 die. The input clock rate can be attained up to 133 MHz, which enables the real time motion estimation for MPEG2 MP＠HL.

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.

We propose the Kernel MUSIC algorithm as an improvement over the conventional MUSIC algorithm. This algorithm is based on the orthogonality between the image and kernel space of an Hermitian mapping constructed from the received data. Spatial smoothing, needed to apply the MUSIC algorithm to coherent signals, is interpreted as constructing procedure of the Hermitian mapping into the subspace spanned by the constituent vectors of the received data. We also propose a new spatial smoothing technique which can remove the redundancy included in the image space of the mapping and discuss that the removal of redundancy is essential for improvement of resolution. By computer simulation, we show advantages of the Kernel MUSIC algorithm over the conventional one, that is, the reduction of processing time and improvement of resolution. Finally, we apply the Kernel MUSIC algorithm to the Laser Microvision, an optical misroscope we are developing, and verify that this algorithm has about two times higher resolution than that of the Fourier transform method.

This paper describes a trial of evaluating the proper characteristics of multiple sound insulatain systems from their output responses contaminated by unknown background noises. The unknown parameters of sound insulation systems are first estimated on the basis of hte linear time series on an intensity scale, describing functionally the input-output relation of the systems. Then, their output probability distributions are predicted when an arbitrary input noise passes through these insulation systems.

This paper proposes a fast timing recovery method with a decision feedback equalizer for baudrate sampling. The proposed method features two special techniques. The first one is for coarse estimation of the sampling phase. Internal signals of the oversampled analog-to-digital converter at different phases are directly taken out for parallel evaluation. The second technique provides fine tuning with a phase-modification stepsize which is adaptively controlled by the residual intersymbol interference. Simulation results by a full-duplex digital transmission system with a multilevel line code show superiority of the proposed method. The coarse timing estimation and the fine tuning reduce 75% and 40% of the time required by the conventional method,respectively. The overall saving in timing recovery is almost 60% over the conventional method. The proposed method could easily be extended to other applications with a decision feedback equalizer.

Internet is evolving to be an integrated service platform for promising future global information infrastructure. New multimedia services, e.g., Video-on-Demand, are expected to be provided on this platform, in conjuction with the existing rich information exchange and sharing services on Internet, such as E-mail, WWW. The paper proposes a service system architecture for Video-on -Demand on Internet, named MINS (Multimedia Internet Navigation System), which is capable of both realtime audio/video transmission and realtime and VCR like plya-out. MINS is composed of 1) servers maintaining both audio/video program lists and contents, and 2) clients retrieving the programs and play-out them. The paper focuses on the communication architecture in MINS, e.g., flow control, realtime media synchronization control, and audio/video session control. The performance of the prototype MINS is also evaluated in the paper.

The evolution of access networks towards Fiber-To-The-Home (FTTH) receives much attention throughout the world because FTTH has a big potential for providing future attractive services. To provide such services in terms of transmission rate, the cable network transition scenario from conventional copper cable network to optical fiber cable one in access field should be clarified. Since network transition scenario deeply depends on the service evolution scenarios (what type of service would appear at a certain time and how many customers move to the other services, for example), this paper firstly categorizes telecommunication services into four kinds of services, i.e. Basic Service (e.g. POTS), Narrowband Symmetrical Service (e.g. NISDN), Broadband Asymmetrical Services (e.g. Video-On-Demand) and Broadband Symmetrical Services (e.g. BISDN) in terms of up-and down-stream transmission rates. Based on this service categorization, it is proposed to formalize service evolution scenarios by a transition pattern of the initial demands and the penetration rates of these services. Then, the evaluation method of network transition scenarios is also proposed in terms of NPV by the service evolution scenarios. It is conclude that the network transition scenarios are deeply related to the amount and behavior of the demands moving to the other services as follows.-The scenario where the new system configuration partly replaces the old one whenever the demands appear for providing higher transmission rate services is applicable, when the demands move little or constantly move to the higher transmission rate services,-The scenario where the new system configuration totally replaces the old one at the early stage for providing all services is applicable, when the demands drastically move to the higher transmission rate services during short period.

This paper first presents the performance analysis of the NACF algorithm. The results show the possibility of the degradation in the convergence speed. To improve the convergence speed, the bias term is introduced into the NACF algorithm and its efficiency is investigated through the computer simulations.

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.

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.

For reducing requirement of large memory and minimizing computation complexity in a large-vocabulary continuous speech recognition system, speech segmentation plays an important role in speech recognition systems. In this paper, we formulate the speech segmentation as a two-phase problem. Phase 1 (frame labeling) involves labeling frames of speech data. Frames are classified into three types: (1) silence, (2) consonant and (3) vowel according to two segmentation features. In phase 2 (syllabic unit segmentation) we apply the concept of transition states to segment continuous speech data into syllabic units based on the labeled frames. The novel class of hyperrectangular composite neural networks (HRCNNs) is used to cluster frames. The HRCNNs integrate the rule-based approach and neural network paradigms, therefore, this special hybrid system may neutralize the disadvantages of each alternative. The parameters of the trained HRCNNs are utilized to extract both crisp and fuzzy classification rules. In our experiments, a database containing continuous reading-rate Mandarin speech recorded from newscast was utilized to illustrate the performance of the proposed speaker independent speech segmentation system. The effectiveness of the proposed segmentation system is confirmed by the experimental results.

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.

In many actual applications of the adaptive filtering, input signals as well as output signals often contain observation noises. Hence, it is necessary to develop an adaptive filtering algorithm to such an errors-in-variables (EIV) model. One solution for identifying the EIV model is a total least squares (TLS) algorithm based on a singular value decomposition of an off-line processing. However, it has not been considered to identify the EIV IIR system using an adaptive TLS algorithm of which stability has been guaranteed during adaptation process. Hence we propose a normalized lattice IIR adaptive filtering algorithm for the TLS parameter estimation. We also show the effectiveness of the proposed algorithm under noisy circumstances through simulations.

In this work, a new structure of M-channel linear-phase paraunitary filter banks is proposed, where M is even. Our proposed structure can be regarded as a modification of the conventional generalized linear-phase lapped orthogonal transforms (GenLOT) based on the discrete cosine transform (DCT). The main purpose of this work is to overcome the limitation of the conventional DCT-based GenLOT, and improve the performance of the fast implementation. It is shown that our proposed fast GenLOT is superior to that of the conventional technique in terms of the coding gain. This work also provides a recursive initialization design procedure so as to avoid insignificant local-minimum solutions in the non-linear optimization processes. In order to verify the significance of our proposed method, several design examples are given. Furthermore, it is shown that the fast implementation can be used to construct M-band linear-phase orthonormal wavelets with regularity.

We present a minimal lattice realization of MIMO linear discrete-time systems which interpolate the desired Markov and covariance parameters. The minimal lattice realization is derived via a recursive construction algorithm based on the state space description and it parametrizes all the interpolants.

High resolution algorithms in sensor arrays lead to accurate results but with expensive eigendecompositions making its use in real-time applications such as mobile communications relatively difficult. In this paper, a trade-off between accuracy and computational load is accomplished through a simplified algorithm which instead of eigendecompositions, uses the robust QR decomposition for which many effcient parallel (systolic, wavefront array) implementations exist. First, a simple detection scheme is presented and, through simulations, is shown to work very well for sufficient SNR, even when signals are coherent. Outputs of the detection process include simultaneously estimates of signals Direction Of Arrivals (DOA's) and a simple beamformer vector resulting in an estimate of the desired signal. Extensive simulations are performed assuming different scenarios of variations in SNR, DOA's leading to discussions on the possibilities and limitations of the proposed solution.

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.

A new ATM switch architecture, named shared multibuffering, features great advantages on memory access speed for a large switch, and overall size of buffer memories to achieve excellent cell-loss performance. We have developed a 622-Mb/s 88 shared multibuffer ATM switch with multicast functions and hierarchical queueing functions to accommodate 156-Mb/s, 622-Mb/s and 2.4-Gb/s interfaces. Implementation of the shared multibuffer ATM switch is described with respect to the four sorts of 0.8-µm BiCMOS LSIs and ATM switch boards. The switch board/type-1, with C1-LSI, allows to accommodate effectively 156-Mb/s and 622-Mb/s interfaces, which is suitable for an ATM access system. The switch board/type-2, with C2-LSI, can provide multicast functions and accommodate a 2.4-Gb/s interface. By using four switch boards, it is possible to apply them to a 2.4-Gb/s ATM loop system.

A 33-cm-Diagonal High-Resolution(1280 1024RGB, which stands for red, green, and blue) TFT-LCD with low, uniform parasitic capacitance between gate electrodes and source/drain electrodes has been developed using Fully Self-Aligned a-Si TFTs. The fabricated TFT-LCD shows no visible seams between block shot exposure regions, even in the display of gray images. In this paper, we describe(1) our full self-alignment technology for the TFTs, including the fabrication process and the technology for reducing OFF current in the TFTs under illumination, (2) SPICE simulation for estimating pixel voltage shift in the fabricated TFT-LCD, and (3) performance results for the fabricated TFT-LCD.

High-resolution algorithms for the detection and estimation of Directions Of Arrival (DOA) such as MUSIC, lead to accurate results but require the computation of the noise-subspace through an expensive covariance matrix eigendecomposition. Adaptive estimators of the noise-subspace can be very useful in a non-stationary environment when the convergence is possible with a few number of snapshots. Some adaptive methods are presented showing that an indirect noise-subspace estimation through a signal subspace estimation can be advantageous both in terms of convergence rate and computation complexity during each update. Some computer simulations examples showing performances are provided.