Most of the existing studies on storing and searching XML documents effectively manipulate each XML document independently. Therefore, techniques for storing XML documents together that have similar meaning or structure are required for efficiency. Also, as a unified access method for various XML storage systems that have different storage forms, studies to integrate the DTD or XML schema of each storage system into one are required, because many XML documents do not have a particular DTD or XML schema, or XML documents can be written in various ways. Therefore, studies on the integration techniques for XML instances are needed. The XML integration technique can be used effectively in the case of constructing a data warehouse for heterogeneous XML storage systems. The proposed integration techniques remove the space duplicated for the same elements in XML documents. The proposed techniques significantly reduce the search time for general queries on the XML documents because it stores the related parts in XML documents close.

This paper proposed a new watermarking algorithm and implementation in hardware, by which the watermarking process and an image compression process can operate in conjunction, in parallel, and/or without degrading the performance of the compression process. The goal of the proposed watermarking scheme is to provide the bases to insist the ownership and to authenticate integrity of the watermark-embedded image by detecting the errors and their positions without the original image (blind watermarking). Our watermarking scheme is to replace the watermark with one or several bit-plane(s) of the DC subband after 2DDWT (2-Dimensional Discrete Wavelet Transform) decomposition which is the basic transformation in DWT-based image compression such as JPEG2000. If more than one bit-plane is involved, the position to embed each watermark bit is randomly selected among the bit-planes by a random number generated with an LFSR (Linear Feedback Shift Register). Experimental results showed that for all the considered attacks except the high compression by JPEG, the error ratios in the extracted watermarks by our algorithm were below 3% and the extracted watermarks were unambiguously recognizable in all the cases. The hardware (FPGA)-implemented result could operate stably in 82 MHz clock frequency. This hardware was merged to DWT-based image compression codec which runs in a real-time in 66 MHz of clock frequency. This resulted in the real-time operation for codec and watermarking together in 66 MHz of clock frequency. The watermarking scheme used 4,037 LABs (24%) of the hardware resource of APEX20KC EP20K400CF672-7 from Altera.

In this paper, a coefficient-parameter reduction method is proposed for invertible deinterlacing with variable coefficients. Invertible deinterlacing, which the authors have developed before, can be used as a preprocess of frame-based motion picture codec, such as Motion-JPEG2000 (MJP2), for interlaced videos. When the conventional field-interleaving is used instead, comb-tooth artifacts appear around edges of moving objects. On the other hand, the invertible deinterlacing technique allows us to suppress the comb-tooth artifacts and also to recover an original picture on demand. As previous works, the authors have developed a variable coefficient scheme with a motion detection filter, which realizes adaptability to local characteristics of given pictures. When applying this deinterlacing technique to an image codec, it is required to send coefficient parameters to receivers for original picture recovery. This work proposes a parameter decimation technique and shows that this reduction approach can be achieved without significant loss of comb-tooth suppression capability and improves the quality at high bit-rate decoding.

Net auctions have been widely utilized with the recent development of the Internet. However, it is a problem that there are too many items for bidders to select the most suitable one. We aim at supporting the bidders on net auctions by automatically generating a table which contains the features of several items for comparison. We construct a system called NTM-Agent (Net auction Text Mining Agent). The system collects web pages of items and extracts the items' features from the pages. After that, it generates a table which contains the extracted features. This research focuses on two problems in the process. The first problem is that if the system collects items automatically, the results contain the items which is different from the items of the user's target. The second problem is that the descriptions in net auctions are not uniform (There are different formats such as sentences, items and tables. The subjects of some sentences are omitted. ). Therefore, it is difficult to extract the information from the descriptions by conventional methods of information extraction. This research proposes methods to solve the problems. For the first problem, NTM-Agent filters the items by correlation rules about the keywords in the titles and the item descriptions. These rules are created semi-automatically by a support tool. For the second problem, NTM-Agent extracts the information by distinguishing the formats. It also learns the feature values from plain examples for the future extraction.

A robust watermarking scheme based on the time-spread echo method is proposed in this letter. The embedding process is achieved by subband decomposition of a host signal and by controlling the amount of distortion, i.e., power of watermark, of each subband according to the Signal to Mask Ratio (SMR) calculated from MPEG psychoacoustic model. The decoding performance and robustness of the proposed method were evaluated.

The current letter extends narrow band (NB) local polynomial approximation (LPA) beamforming to wide band (WB) rapidly moving sources. Instead of the conventional beamformer weight in NB LPA, the proposed method adopts the steered minimum variance (STMV) method that can achieve a high resolution with short time observations. The performance of the proposed algorithm is demonstrated via computer simulations.

Recently, Ma and Chen proposed a new authenticated encryption scheme with public verifiability. The signer can generate a signature with message recovery for a specified recipient. With a dispute, the recipient has ability to convert the signature into an ordinary one that can be verified by anyone without divulging her/his private key and the message. However, we point out that any adversary can forge a converted signature in this article.

An algorithm is developed for augmenting a real video with virtual graphics objects without computing Euclidean information. For this, we design a method of specifying the virtual camera that performs Euclidean orthographic projection in recovered affine space. In addition, our method has the capability of generating views of objects shaded by virtual light sources. Our novel formulation and experimental results are presented.

This paper proposes a novel method of identifying the design parameters for a practical implementation of the fair queueing discipline, which is capable of class-level delay control. The notion of class weight is introduced at first, and then the session weights are determined. This two-phase approach is favorable in terms of the scalability;that is, the overall complexity is dependent upon the number of classes only. We propose a packet scheduler referred to as the DPS (Delay-centric Processor Sharing) scheme which employs those design parameters to deliver class-wise delay bound services. The associated admission policy for delay guarantee is also derived. System analysis and derivation of the parameters have their origins in the understanding of the so-called system equation, which describes the dynamics of the class-level service share. The proposed design parameters are QoS-aware in that they are consistently refined depending on the system status. Several numerical and simulation results show that the DPS scheme is advantageous over other ones in terms of both resource efficiency and the robustness. Concerning the scalability, we show that an alternative tagging process of the DPS scheme is implementable with O(1) complexity with no significant degradation in delay performance.

This letter discusses the performance of online SNR estimation including fading parameter estimation for parallel combinatorial SS (PC/SS) systems. The PC/SS systems are partial-code-parallel multicode SS systems, which have high-rate data transmission capability. Nakagami-m distribution is assumed as fading channel model to cover a wide range of fading conditions. The SNR and fading parameter estimation considered in this letter is based on only a statistical ratio of correlator outputs at the receiver. Numerical results show that SNR estimation performance with fading parameter estimation is close to the one in the case of perfect fading parameter information, if the number of transmitting PN codes is less than a half of assigned PN codes.

There has recently been considerable interest in research on wearable non-grounded force display. However, there have been no developments for the communication of nonverbal information (ex. tennis and golf swing). We propose a small and lightweight wearable force display to present motion timing and direction. The display outputs a torque using rotational moment and mechanical brakes. We explain the principle of this device, and describe an actual measurement of the torque and torque sensitivity experiments.

Integral equation method is used to compute three-dimension-structure capacitance in this paper. Since some multi-conductor structures present regular periodic property, the periodic cell is used to reduce the computational domain with adding appropriate magnetic and electric walls. The periodic Green's function in the integral equation method is represented in the form of infinite series with slow convergence. In this paper, Shanks transformation is used to accelerate the convergence. Numerical examples show that the proposed method is accurate with a much higher efficiency in capacitance extraction for 3-D periodic structures.

This paper presents the improvement of the transient response and stability for a two-stage DC-DC converter by removing the output inductor. The conventional two-stage converter consists of a buck converter used as the first stage and a half-bridge converter used as the second stage. The proposed circuit topology removing the output inductor and the conventional topology are compared. Removing the output inductor results in the system-order reduction of the transfer function. As a result, the stability is improved, and the crossover frequency of the open-loop transfer function becomes higher. The effectiveness of the proposed circuit topology was experimentally confirmed.

Dynamic non-linearities are of more importance in highly-linear high-speed applications such as software radios. In this paper, a fully-analytical approach to estimate the statistics of dynamic non-linearity parameters of pipeline analog-to-digital converters (ADCs) in the presence of circuit non-idealities is presented. These imperfections include the capacitor mismatches and the non-idealities in the operational amplifiers (op-amps). The most two important ADC dynamic non-linearity parameters, the spurious-free dynamic range (SFDR) and the signal-to-noise-and-distortion ratio (SNDR) are quantified here and closed-form formulas are presented. These formulas are useful for design automation as well as hand calculations of highly-linear pipeline ADCs. Behavioral simulations are presented to show the accuracy of the proposed equations.

This paper investigates the relations between IP network performances and the speech quality of the Voice over IP (VoIP) service through extensive experiments on a test bed network. The aim is to establish an effective and practical methodology for telecommunications operators to manage the quality of VoIP service via the management of IP network performances under their control. As IP network performances, utilization of the bottleneck link in the test bed and the following statistical factors of VoIP packets are examined: the standard deviation of delay variations (jitters), the standard deviation of packet interarrival times, and the packet loss ratio. On the other hand, VoIP speech quality is monitored as the Perceptual Evaluation of Speech Quality (PESQ). To investigate the relations under various network conditions, the experiments are performed by varying the following network related parameters of the test bed: the bandwidth of the bottleneck link, the size of the bottleneck buffer, the propagation delay, and the average of the data sizes transmitted as background data traffic. Statistical analyses of the experimental results suggest that managing the standard deviation of jitters in a network serves as a promising methodology, because its close relation to VoIP speech quality possesses robustness to changes in the network conditions. The robustness makes it practically useful since telecommunications operators can apply it to their networks, which are subject to change. The findings in this paper have opened up new visions for telecommunications operators to manage the Quality of Service (QoS) of VoIP service.

In General Packet Radio Service (GPRS), combined mobility management is defined to efficiently perform both GSM and GPRS mobility management in a combined manner. In this letter, an enhanced scheme for combined mobility management based on a new mobile station (MS) state model is proposed. The steady state probabilities of the proposed MS state model are derived and the performance of the proposed scheme is analyzed. The results show that the proposed scheme outperforms the conventional scheme, especially for low mobility MSs.

This paper describes a 2.4-GHz downconverter that runs on a 1-V supply. The downconverter integrates an LNA, a quadrature mixer, a complex channel-select band-pass filter (BPF), a limiting amplifier, and a frequency doubler using 0.2-µm CMOS/SOI technology. The frequency doubler doubles the frequency deviation of FM signals as well as the frequency itself, which in turn doubles the modulation index. This improves the sensitivity of FM demodulation. The power consumption of the downconverter is 23 mW with a 1-V power supply. A bit-error-rate (BER) measurement using the downconverter and a demodulation IC shows -76.5-dBm sensitivity at a 0.1% BER.

The accuracy of channel estimation significantly affects the performance of coherent rake receiver in DS-CDMA systems. It is desirable for improved channel estimation to employ a channel estimation filter (CEF) whose bandwidth is adjustable to the channel condition. In this paper, we consider the use of moving average (MA) FIR filters as the CEF since it is simple to implement and can provide relatively good receiver performance. First, we optimize the tap size of the MA FIR CEF so as to minimize the mean squared error of the estimated channel impulse response. For practical applications, we propose a low-complexity adaptive channel estimator (ACE), where the tap size of the MA FIR CEF is adjusted based on the estimated channel condition by exploiting the correlation characteristics of the received pilot signal. Numerical results show that the use of the proposed ACE can provide the receiver performance comparable to that of Wiener CEF without exact a priori information on the operating condition.

We have proposed a new low-IF transceiver architecture to simultaneously achieve both a small chip area and good minimum input sensitivity. The distinctive point of the receiver architecture is that we replace the complicated high-order analog filter for channel selection with the combination of a simple low-order analog filter and a sharp digital band-pass filter. We also proposed a high-speed convergence AGC (automatic gain controller) and a demodulation block to realize the proposed digital architecture. For the transceiver, we further reduce the chip area by applying a new form of direct modulation for the VCO. Since conventional VCO direct modulation tends to suffer from variation of the modulation index with frequency, we have developed a new compensation technique that minimizes this variation, and designed the low-phase noise VCO with a new biasing method to achieve large PSRR (power-supply rejection ratio) for oscillation frequency. The test chip was fabricated in 0.35-µm BiCMOS. The chip size was 3 3 mm2; this very small area was realized by the advantages of the proposed transceiver architecture. The transceiver also achieved good minimum input sensitivity of -85 dBm and showed interference performance that satisfied the requirements of the Bluetooth standard.

There is a fatal difference in obtaining information between sighted people and the blind. Screen reading technology assists blind people in accessing digital documents by themselves helping to bridge such gap. However, these days they are becoming much more visual using various types of visual effects for sighted people to explore the information intuitively at a glance. It is very hard to convey visual effects non-visually and intuitively while retaining the original effects. In addition, it takes a long time to explore the information, since blind people use the keyboard for exploration, while sighted people use eye movement. This research aims at improving the non-visual exploration interface and improving the quality of non-visual information. Therefore, TAJODA (tactile jog dial interface) was proposed to solve these problems. It presents verbal information (text information) in the form of speech, while nonverbal information (visual effects) is represented in the form of tactile sensations. It uses a jog dial as an exploration device, which makes it possible to explore forward or backward intuitively in the speech information by spinning the jog dial clockwise or counterclockwise. It also integrates a tactile device to represent visual effects non-visually. Both speech and tactile information can be synchronized with the dial movements. The speed of spinning the dial affects the speech rate. The main part of this paper describes an experimental evaluation of the effectiveness of the proposed TAJODA interface. The experimental system used a preprocessed recorded human voice as test data. The training sessions showed that it was easy to learn how to use TAJODA. The comparison test session clearly showed that the subjects could perform the comparison task using TAJODA significantly faster (2.4 times faster) than with the comparison method that is closest to the existing screen reading function. Through this experiment, our results showed that TAJODA can drastically improve the non-visual exploration interface.