We report a specially designed encoding technique for sign language video sequences supposing that the technique is for sign telecommunication such as that using mobile videophones with a low bitrate. The technique is composed of three methods: gradient coding, precedence macroblock coding, and not-coded coding. These methods are based on the idea to distribute a certain number of bits for each macroblock according to the evaluation of importance of parts of the picture. They were implemented on a computer and encoded data of a short clip of sign language dialogue was evaluated by deaf subjects. As a result, the efficiency of the technique was confirmed.

Assistive technology (AT) is becoming increasingly important for improving the mobility and language learning capabilities of persons with disabilities, thus enabling them to function independently and to improve their social opportunities. The Morse code has been shown to be a valuable tool in assistive technology, augmentative and alternative communication, and rehabilitation for people with neuromuscular diseases such as amyotrophic lateral sclerosis, multiple sclerosis, and muscular dystrophy. In this paper, we designed and implemented a wireless environmental control aid system using the Morse code as an adapted access communication tool, which includes three types of switch: single-switch, double-switch, and six-switch types. People with disabilities can easily control all types of electronic appliance without restrictions owing to spatial arrangements using a signal transmission based on radio frequency (RF). Experimental results revealed that three participants with disabilities were able to gain access to electronic facilities after six weeks of practice with the new system.

The fast handover protocol adopted in a IPv6 hierarchical structure provides a seamless handover in wireless IP networks by minimizing the handover latency. To reduce the handover latency, the fast handover uses anticipation based on layer 2 trigger. Nonetheless, a mobile node can still lose its connection with the old link during the fast handover procedures. Accordingly, this paper analyzes the handover latency and packet delivery costs associated with fast handover failure cases based on a timing diagram.

This paper proposes a simple and efficient method to numerically obtain the mapping degree deg(f, 0, B) of a C1 map f : Rn → Rn at a regular value 0 relative to a bounded open subset B ⊂ Rn. For practical application, this method adopts Aberth's algorithm which does not require computation of derivatives and determinants, and reduces the computational cost with two additional procedures, namely preconditioning using the coordinate transformation and pruning using Krawczyk's method. Numerical examples show that the proposed method gives the mapping degree with 2n+1 operations using interval arithmetic.

This paper proposes a novel UWB ranging scheme employing 1-bit ADCs and analog window bank for energy collection. For an appropriate 1-bit ADC process DC offset is exploited and removed via performing analog low pass filter. To improve ranging accuracy in presence of ambiguity, dual overlapped window banks designated as primary and auxiliary windows are utilized. Corresponding to the proposed ranging scheme, its performance is verified by conducting simulations in two types of channel conditions. The simulation results show that the proposed ranging scheme performs well even in condensed multipath environment and low SNR situation.

The explosive growth of wireless network users and the existence of various wireless services have demanded high throughput as well as user's quality-of-service (QoS) guarantees. In accordance with, this paper proposes a novel resource allocation scheme improving both the capability of QoS-provisioning for multiple users and the overall data throughput. Towards this, the modified resource allocation technique combined with the modified largest weighted delay first (M-LWDF) scheme will be exploited upon considering statistical channel behavior as well as real time queuing analysis connected to resource allocation. In order to verify the validity of the proposed resource allocation scheme, the time division multiple access (TDMA) system will be considered as a target application. The simulation results confirm that the proposed scheme gives rise to superior performance in a way of showing results of several performance measures under time-varying wireless fading channel.

This paper proposes a low power scan test scheme and formulates a problem based on this scheme. In this scheme the flip-flops are grouped into N scan chains. At any time, only one scan chain is active during scan test. Therefore, both average power and peak power are reduced compared with conventional full scan test methodology. This paper also proposes a tabu search-based approach to minimize test application time. In this approach we handle the information during deterministic test efficiently. Experimental results demonstrate that this approach drastically reduces both average power and peak power dissipation at a little longer test application time on various benchmark circuits.

A great deal of research has been made to model the vagueness and uncertainty in information retrieval. One such research is fuzzy ranking models, which have been showing their superior performance in handling the uncertainty involved in the retrieval process. However, these conventional fuzzy ranking models have a limited ability to incorporate the user preference when calculating the rank of documents. To address this issue, in this study we develop a new fuzzy ranking model based on the user preference. Through the experiments on the TREC-2 collection of Wall Street Journal documents, we show that the proposed method outperforms the conventional fuzzy ranking models.

An unsupervised adaptive signal processing method of principal components analysis (PCA) neural networks (NN) based on signal eigen-analysis is proposed to permit the eigenstructure analysis of lower signal to noise ratios (SNR) direct sequence spread spectrum (DS) signals. The objective of eigenstructure analysis is to estimate the pseudo noise (PN) of DS signals blindly. The received signal is firstly sampled and divided into non-overlapping signal vectors according to a temporal window, which duration is two periods of PN sequence. Then an autocorrelation matrix is computed and accumulated by these signal vectors one by one. Lastly, the PN sequence can be estimated by the principal eigenvector of autocorrelation matrix. Since the duration of temporal window is two periods of PN sequence, the PN sequence can be reconstructed by the first principal eigenvector only. Additionally, the eigen-analysis method becomes inefficient when the estimated PN sequence is long. We can use an unsupervised adaptive method of PCA NN to realize the PN sequence estimation from lower SNR input DS-SS signals effectively.

High-resolution FMCW reflectometry is often realized by sampling the beat signal with a clock signal generated from an auxiliary interferometer. The drawback of this system is that the measurement range is limited to less than half of the optical path difference of the auxiliary interferometer to satisfy the Sampling theorem. We propose and demonstrate a method to extend the measurement range of the system. The clock signal gerenerated from the auxiliary interferometer is electronically frequency-multipled by using a PLL circuit. The measurement range is experimentally extended by a factor of 20 while keeping high spatial resolution, and is theoretically extended by a factor of 128. The advantage of the proposed system is that the optical path difference of the auxiliary interferometer can be kept short, which is very effective for obtaining the stable and low time-jitter clock signal.

In this letter two noise-predictive equalization schemes, which are based on zero-forcing (ZF) and minimum mean square error (MMSE) criteria, are presented for unique-word (UW) based single-carrier systems. The correlation properties of the noises in the outputs of the frequency domain equalizer are exploited to predict and cancel the noise contained in the estimation of data. Theoretical analyses show that both of the proposed techniques perform better than the conventional frequency domain equalizers. Simulation results have confirmed the significant performance improvement they could achieve.

The RC mismatch among S/H stages for time-interleaved ADCs causes a phase error and a gain error and the phase error is dominant. The paper points out that clock skew and the phase error caused by the RC mismatch have similar effects on the sampling error and then can be compensated with the clock skew compensation. Simulation results agree well with the theoretical analysis. With the phase error compensation of RC mismatch, the SNDR in 14b ADC can be improved by more than 15 dB in the case that the bandwidth of S/H circuits is 3 times the sampling frequency. This paper also proposes a method of clock skew and RC mismatch compensation in time-interleaved sample-and-hold (S/H) circuits by sampling clock phase adjusting.

We observe the inter-carrier interference (ICI) caused by channel variation of the fading in time domain in orthogonal frequency division multiplexing (OFDM) systems. This observation allows us to propose simple two-stage equalizer to minimize the ICI. Simulation results show that the bit error rate (BER) performance of the proposed equalizer with much reduced complexity is comparable with that of the classical frequency domain linear minimum mean squared error (MMSE) equalizer.

In this paper, we propose an interactive instruction system named IMPRESSION, which allows performance of interactive presentations using multimedia educational materials in class. In recent years, although many practices of educational methodology with information technology and presentation tools using multimedia resources as educational materials have come into common use, instructors can only present such materials in a slide-sheet form through the use of such presentation tools in class. Therefore, instructors can neither do formative evaluations nor can they present suitable materials according to students' reactions in class. Our proposed methodology employs a scenario-based approach in a double loop instructional design process to overcome such problems. Instructors design an instructional plan as a scenario, and subsequently implement and modify the plan through formative evaluation during the class. They then conduct a summative evaluation based on planned and implemented instructions for redesign. To realize our methodology, in this paper we propose and design an instruction system that provides functions to select and present multimedia materials interactively provided on the Internet during the class; we then record these instructions. After implementing it, we confirmed that we can conduct the class flexibly based on our methodology through its practical use in an actual classroom environment.

According to the Gestalt principals, this paper presents a recognition method of grouping areas in trademark images modeling features for measuring the attraction degree between couples of image components. This investigation would be used for content-based image retrieval from the view of mirroring human perception for images. Depending on variability in human perception for trademark images, the proposed method finds grouping areas by calculating Mahalanobis distance with the features to every combination of two components in images. The features are extracted from every combination of two components in images, and the features represent proximity, shape similarity, and closure between two components. In addition, changing combination of the features, plural grouping patterns are output. Besides, this paper shows the efficiency and limits of the proposed method from experimental results. In the experiments, 104 participants have perceived grouping patterns to 74 trademark images and the human perceptions have been compared with outputs by the proposed method for the 74 images.

In this paper, ultra-wideband (UWB) multiple access systems are introduced by using direct-sequence (DS) and hybrid direct-sequence time-hopping (DS/TH) code division multiple access (CDMA) that use arbitrary chip-duty of the spreading sequences. The bit error probabilities are presented. First of all, the variances of the multiple access interference are developed by investigating the collision properties of the signals. Afterward, various approximations are applied. The standard Gaussian approximation (SGA) for the DS system is shown to become extremely optimistic as the chip-duty becomes low. Though the hybrid system performs better, the SGA still remains optimistic. To obtain accurate results, Holtzman's simplified improved Gaussian approximation (SIGA) and Morrow and Lehnert's improved Gaussian approximation (IGA) are used. A shortcoming of the SIGA is rediscovered that renders it unusable for low-duty DS systems, especially, at high signal-to-noise ratio. However, for the hybrid system, the SIGA works as an excellent tool. The IGA is used to get accurate results for the low-duty DS systems. It is shown that lowering of chip-duty by keeping chip rate and chip length unchanged improves performance for asynchronous DS and both asynchronous and synchronous hybrid systems. However, under the same processing gain, a high-duty system performs better than a low-duty system. Performance of synchronous DS system remains independent of chip-duty.

The purpose of the study is to obtain the automatic and optimal matching between a motion-measurement device such as a data glove and an output device such as a dexterous robot hand, where there are many differences in the numbers of degree of freedom, sensor and actuator positions, and data format, by means of motion imitation by the humans. Through the algorithm proposed here, a system engineer or user need no labor of determining the values of gains and parameters to be used. In the system, a subject with data glove imitated the same motion with a dexterous robot hand which was moving according to a certain mathematical function. Autoregressive models were adapted to the matching, where each joint angle in the robot and data glove data of the human were used as object and explanatory variables respectively. The partial regression coefficients were estimated by means of singular value decomposition with a system-noise reduction algorithm utilizing statistical properties. The experimental results showed that the robot hand was controlled with high accuracy with small delay, suggesting that the method proposed in this study is proper and easy way and is adaptive to many other systems between a pair of motion-measurement device and output device.

A CMOS circuit is proposed which takes multiple analog input currents and extracts minimum input current at the output. It is very fast and requires no subtraction from the constant current source. It exhibits O(N) complexity and uses only 4N MOS transistors where N is the number of system inputs. This circuit consumes very little power and very small area. The substrate bias affects the threshold voltage of transistors and improves performance of the structure.

This paper describes the design of a small-offset 12-bit CMOS charge-redistribution DAC using a weighted-mean flip-around sample-and-hold circuit (S/H). Flip-around S/H topology can realize small-offset characteristics, and it is effective to reduce power dissipation and chip area because independent feedback capacitors are not necessary. In this DAC the small-offset characteristic remains not only in amplification phase but also in sampling phase with the circuit technique. The design of 1.8 V, 50 MS/s fully differential DAC with output swing of 2 Vp-p has very small offset of 100 µV for the reset switch mismatch of 2%. A technique to improve dynamic performance measured by SFDR using damping resistors and switches at the output stage is also presented. The designed 12-bit DAC with 0.25 µm CMOS technology has low-power dissipation of 35 mW at 50 MS/s.

Two congestion control schemes designed specifically to handle changes in the datalink interface of a mobile host are presented. The future mobile environment is expected to involve multimode connectivity to the Internet and dynamic switching of the connection mode depending on network conditions. The conventional Transmission Control Protocol (TCP), however, is unable to maintain stable and efficient throughput across such interface changes. The two main issues are the handling of the change in host Internet Protocol (IP) address, and the reliability and continuity of TCP flow when the datalink interface changes. Although existing architectures addressing the first issue have already been proposed, the problem of congestion control remains. In this paper, considering a large change in bandwidth when the datalink interface changes, two new schemes to address these issues are proposed. The first scheme, Immediate Expiration of Timeout Timer, detects interface changes and begins retransmission immediately without waiting for a retransmission timeout as in existing architectures. The second scheme, Bandwidth-Aware Slow Start Threshold, detects the interface change and estimates the new bandwidth so as to set an appropriate slow start threshold for retransmission. Through simulations, the proposed schemes are demonstrated to provide marked improvements in performance over existing architectures.