A time-domain equalization (TEQ) algorithm is presented to shorten the effective channel impulse response to increase the transmission efficiency of the 54 Mbps IEEE 802.11a orthogonal frequency division multiplexing (OFDM) system. In solving the linear equation Aw = B for the optimum TEQ coefficients, A is shown to be Hermitian and positive definite. The LDLT and LU decompositions are used to factorize A to reduce the computational complexity. Simulation results show high performance gains at a data rate of 54 Mbps with moderate orders of TEQ finite impulse response (FIR) filter. The design and implementation of the algorithm in field programmable gate array (FPGA) are also presented. The regularities among the elements of A are exploited to reduce hardware complexity. The LDLT and LU decompositions are combined in hardware design to find the TEQ coefficients in less than 4 µs. To compensate the effective channel impulse response, a radix-4 pipeline fast Fourier transform (FFT) is implemented in performing zero forcing equalization. The hardware implementation information is provided and simulation results are compared to mathematical values to verify the functionalities of the chips running at 54 Mbps.

Interval arithmetic is able to be applied when we include the ranges of various functions. When we include them applying the interval arithmetic, the serious problem that the widths of the range inclusions increase extremely exists. In range inclusion of polynomials particularly, Horner's method and Alefeld's method are well known as the conventional methods which mitigates this problem. The purpose of this paper is to propose the new methods which are able to mitigate this problem more efficiently than the conventional methods. And in this paper, we show and compare the efficiencies of the new methods by some numerical examples.

This paper investigates modified random timers based on uniform and exponentially distributed timers for feedback scalability for large groups. We observe the widely-used probability distribution functions and propose new ones that are aware of network delays. The awareness of network delays of our proposed modified p.d.fs proves to be able to achieve lower expected number of messages compared to the original ones given that the parameters are optimized for the network variables: the number of receivers, and the network delay. In our analysis we derive an equation to estimate the optimized parameter based on these network variables. We also simulate the p.d.fs for heterogenous network delays and find that each receiver only needs to be aware of its network delay.

When a feedback bridging fault occurs in a combinational circuit and it is activated, logical oscillation may occur in the circuit. In this paper, some electrical conditions are proposed to identify whether a feedback bridging fault occurs logical oscillation. Also, it is proposed how to estimate the oscillation frequency. They are based on piece linearlized models and do not require circuit simulation of large size of circuits. They are evaluated by some experiments. In the experiments, all of the feedback bridging faults occurring logical oscillation are identified. Also, oscillation frequencies larger than the ones obtained by SPICE simulation are derived by the proposed estimation method in the experiments. It promises us that the methods will be used for identifying such bridging faults and estimating the oscillation frequencies.

The partially rotational scan (PRS) technique greatly reduces the amount of data needed for n-detection testing. It also enables at-speed testing using low-speed testers. We designed tester intellectual properties (tester IP) with PRS for Viper and COMET II processors. When PRS was applied to a Viper processor, we obtained test data that provided the same fault coverage as with a set of automatic test pattern generation (ATPG) test vectors, although the amount of test data was 16% that of the ATPG. When the PRS technique was applied to a COMET II processor with full-scan design, we obtained test data that provided the same fault coverage as with a set of ATPG test vectors, although the amount of test data was 10% that of the ATPG. We also estimated hardware overhead and test time.

In this paper, we propose a speech coding translation scheme by transferring coding parameters between GSM half rate and G.729 coders. Compared to the conventional decode-then-encode (DTE) scheme, the proposed parameter conversions provide speech interoperability between mobile and IP networks with reducing computational complexity and coding delay. Simulation results show that the proposed methods can reduce about 30% computational load and coding delay acquired in the target encoders and achieve almost imperceptible degradation in performance.

This paper proposes a new decision feedback decoding scheme for Alamouti-based space-time block coding (STBC) transmission over time-selective fading channels. In wireless channels, time-selective fading effects arise mainly due to Doppler shift and carrier frequency offset. Modelling the time-selective fading channels as the first-order Gauss-Markov processes, we use recursive algorithms such as Kalman filtering, LMS and RLS algorithms for channel tracking. The proposed scheme consists of the symbol decoding stage and channel tracking algorithms. Computer simulations confirm that the proposed scheme shows the better performance and robustness to time-selectivity.

Rectangle search is a well-known packet classification scheme which is based on multiple hash accesses for different filter length. It shows good scalability with respect to the number of filters; however, the performance is not fast enough to fulfill the high-speed requirement of packet classification. In this paper, we propose a lookahead caching which can significantly improve the performance of hash-based algorithm. The basic idea is to filter out the un-matched probing case by using dual-hash architecture. The experimental results indicate that the proposed scheme can improve the performance by the factor of two for the 2-dimension (source prefix, destination prefix) filter database.

Although many researchers have verified the superiority of Support Vector Machine (SVM) on text categorization tasks, some recent papers have reported much lower performance of SVM based text categorization methods when focusing on all types of parts of speech (POS) as input words and treating large numbers of training documents. This was caused by the overfitting problem that SVM sometimes selected unsuitable support vectors for each category in the training set. To avoid the overfitting problem, we propose a two step text categorization method with a variable cascaded feature selection (VCFS) using SVM. VCFS method selects a pair of the best number of words and the best POS combination for each category at each step of the cascade. We made use of the difference of words with the highest mutual information for each category on each POS combination. Through the experiments, we confirmed the validation of VCFS method compared with other SVM based text categorization methods, since our results showed that the macro-averaged F1 measure (64.8%) of VCFS method was significantly better than any reported F1 measures, though the micro-averaged F1 measure (85.4%) of VCFS method was similar to them.

Self-stabilization is a theoretical framework of non-masking fault-tolerant distributed algorithms. In this paper, we investigate the Steiner tree problem in distributed systems, and propose a self-stabilizing heuristic solution to the problem. Our algorithm is constructed by four layered modules (sub-algorithms): construction of a shortest path forest, transformation of the network, construction of a minimum spanning tree, and pruning unnecessary links and processes. Competitiveness is 2(1-1/l), where l is the number of leaves of optimal solution.

A total coloring of a graph G is to color all vertices and edges of G so that no two adjacent or incident elements receive the same color. Let C be a set of colors, and let ω be a cost function which assigns to each color c in C a real number ω(c) as a cost of c. A total coloring f of G is called an optimal total coloring if the sum of costs ω(f(x)) of colors f(x) assigned to all vertices and edges x is as small as possible. In this paper, we give an algorithm to find an optimal total coloring of any tree T in time O(nΔ3) where n is the number of vertices in T and Δ is the maximum degree of T.

Our study focuses on an evaluation of a novel speech processing strategy for multi-channel cochlear implant speech processors. Stimulation pulse trains for the Nucleus 24CI speech processor were generated in a way different from the speech processing strategies implemented in this processor. The distinctive features of the novel strategy are: 1) electrode stimulation order driven by location of maximum instantaneous frequency amplitude; 2) variable stimulation rates on electrodes; 3) variable number of selected channels within a cycle of signal processing schema. Within-subject designed tests on Japanese initial, medial and final consonants in CV, VCV and CV/N context tokens were carried out with cochlear implant patients using the Cochlear ACETM strategy, and results were compared with those of normal hearing listeners. Results of the initial and medial consonant tests showed significantly better performance with the novel strategy than with the ACE strategy for both the cochlear implant and normal hearing listener groups. Results of the final consonant tests showed a slightly better performance with the ACE strategy for cochlear implant listeners while showing a slightly better performance with the novel strategy for normal hearing listeners.

In this paper, we consider a problem of global stabilization of a class of nonlinear systems which are approximately feedback linearizable. We propose a control law with the gain-scaling factor and analytically show the robust aspect of approximate feedback linearization in a more general framework.

IEEE 802.11 standard and the enhanced amendments have defined several transmission rates for mobile stations to transmit and receive data frames. With the characteristic of modulation schemes, a higher network throughput can be expected by using higher level modulation scheme, but frame error probability will also become higher. Doubtlessly, it is an open issue of selecting a proper modulation scheme for a pair of mobile stations in time-varying indoor environment. This paper proposes a safe multiple access-rates transmission (SMART) scheme to enhance the reliability of data transmission in IEEE 802.11 multi-rate infrastructure wireless networks. SMART scheme provides reliable transmission by reserving a retransmission period which immediately follows the transmitted frame and is estimated by a lower transmission rate according to the transmitted frame. The performance of SMART scheme is evaluated by analytical model and simulation experiment, which show that the proposed scheme is significantly better than the IEEE 802.11 standard under the real environment with asymmetric traffic load. An enhanced SMART (ESMART) scheme is also proposed especially to improve bandwidth waste for long packet length at access point (AP). The performance measurements, such as goodput, throughput, average access delay, and service rate, are obtained and compared for the different schemes.

This paper describes feasibility of a proposed fixed wireless access system with CDMA technology. The system adopts a primary modulation of 16 QAM and the same frequency allocation in all cells to improve spectral efficiency. The system capacity is 1 Gbps per cell within 120 MHz bandwidth. The number of available orthogonal codes corresponds to the orthogonal code length in the system. All subscribers can attain an error free condition with output power control in the presence of inter-cell interference. The following two items are considered to examine the proposed system feasibility. 1) A test modem is fabricated, and a back-to-back modem BER performance is measured. An inter-symbol interference (ISI) level of the modem is estimated with the measured performance. 2) A computer simulation of down-link power control is carried out considering inter-cell interference and impairment factors of the power control such as intra-sector interference caused by the ISI and limited ranges of total and relative output power controls. The simulation results show that the proposed system would be feasible because the obtained power penalties caused by the above impairment factors are negligible.

The high performance method for analyzing the time evolution of the shielding current density in the high Tc superconductor (HTS) has been investigated. After discretized by using the finite element method and the backward Euler method, the initial-boundary-value problem of the governing equations of the shielding current density is transformed to the problem in which the nonlinear algebraic equations are solved at each time step. When the deaccelerated Newton method (DNM) is applied to the solution of the equations, a decrease in the relaxation factor will not always ensure the convergence of iterations. For this reason, the DNM is modified so that the residual norm may decrease monotonously with the iteration number. The modified method is called the adaptively deaccelerated Newton method (ADNM). Although the vector function is evaluated several times at each cycle in the ADNM, the CPU time required for the ADNM is diminished considerably as compared with that for the DNM. This result indicates that the ADNM is suitable for calculating the shielding current density. The numerical code for analyzing the shielding current density has been developed on the basis of the ADNM and, as an application of the code, the magnetic shielding performance of an axisymmetric HTS plate has been analyzed.

We propose a new method for Depth from Defocus (DFD) using wavelet transform. Most of the existing DFD methods use inverse filtering in a transform domain to determine the measure of defocus. These methods suffer from inaccuracies in finding the frequency domain representation due to windowing and border effects. The proposed method uses wavelets that allow performing both the local analysis and windowing with variable-sized regions for images with varying textural properties. Experimental results show that the proposed method gives more accurate depth maps than the previous methods.

In this paper, we propose a new method of automatic speech summarization for each utterance, where a set of words that maximizes a summarization score is extracted from automatic speech transcriptions. The summarization score indicates the appropriateness of summarized sentences. This extraction is achieved by using a dynamic programming technique according to a target summarization ratio. This ratio is the number of characters/words in the summarized sentence divided by the number of characters/words in the original sentence. The extracted set of words is then connected to build a summarized sentence. The summarization score consists of a word significance measure, linguistic likelihood, and a confidence measure. This paper also proposes a new method of measuring summarization accuracy based on a word network expressing manual summarization results. The summarization accuracy of each automatic summarization is calculated by comparing it with the most similar word string in the network. Japanese broadcast-news speech, transcribed using a large-vocabulary continuous-speech recognition (LVCSR) system, is summarized and evaluated using our proposed method with 20, 40, 60, 70 and 80% summarization ratios. Experimental results reveal that the proposed method can effectively extract relatively important information by removing redundant or irrelevant information.

This paper presents a novel wavelet compression technique to increase compression of images. Based on zerotree entropy coding method, this technique initially uses only two symbols (significant and zerotree) to compress image data for each level. Additionally, sign bit is used for newly significant coefficients to indicate them being positive or negative. Contrary to isolated zero symbols used in conventional zerotree algorithms, the proposed algorithm changes them to significant coefficients and saves its location, they are then treated just like other significant coefficients. This is done to decrease number of symbols and hence, decrease number of bits to represent the symbols used. In the end, algorithm indicates isolated zero coordinates that are used to change the value back to original during reconstruction. Noticeably high compression ratio is achieved for most of the images, without changing image quality.

Among the many scheduling algorithms which can guarantee delay bounds of sessions, the EDF scheduler with regulators has received wide attention since it can admit a very high number of sessions. However, due to regulators, EDF with regulators has difficulty in scalable implementation. To implement a regulator, a buffer and a timer are needed to restrict the incoming traffics. Given N number of sessions, N regulators are required. Moreover, due to regulators, the entire scheduling algorithm is not work-conserving. To enforce work-conserveness, it is known in the literature that additional buffers and a complex mechanism are required to bypass regulators. Thus, scalable implementation becomes much more difficult in the case of the work-conserving EDF with regulators. In this paper, however, we show that the buffers and timers used to implement regulators are unnecessary to guarantee delay bounds of admitted sessions in fixed-sized packet networks. Then, we can remove those unnecessary buffers and timers. By the removal, the resulting scheduling algorithm can be implemented in a scalable way and becomes work-conserving for free.