2.4 GHz-band wireless LAN system based on a new standard, IEEE 802.11g, has been taking a great attention as it provides the attractive features such as low cost, unlicensed spectrum use, and high speed transmission rate up to 54 Mbps. However, 2.4 GHz radio frequency band is also used for Industrial, Scientific and Medical (ISM) devices such as microwave ovens, and the man-made noise leaked from ISM devices is known to be one of the major causes of the degradation in the performance of wireless communications systems using 2.4 GHz radio frequency band. In this paper, we evaluate the bit error rate (BER) and the throughput performances of WLAN system based on IEEE 802.11g standard (IEEE 802.11g WLAN system) under man-made noise environment, and discuss the effect of man-made noise on the performance of IEEE 802.11g WLAN system. Numerical results show that the BER and the throughput performances of IEEE 802.11g WLAN system are much degraded by the influence of man-made noise.

In this letter, we propose an efficient closed-loop transmit (Tx) diversity scheme that works well for high mobility as well as low mobility. The proposed scheme exploits a quantized weight vector codebook designed by separating it into gain and phase codebooks. Simulation results reveal that the proposed scheme can provide a significant advantage in both complexity and flexibility over conventional methods.

The increasing number of channels in dense wavelength division multiplexing (DWDM) systems has led to the need for wiring involving a large number of optical fibers in the system racks. We have developed a novel scalable optical fiber wiring system designed to realize as many as 10,000-fiber shuffled interconnections without fiber congestion. We propose a scheme for constructing a large-scale shuffler capable of permuting interconnected fibers that employs plural optical fiber sheets, and for arranging optical fibers without congestion in racks. We constructed a 16,384-fiber shuffler system with sixty-four 256-fiber shuffler sheets and 16-fiber fiber physical contact (FPC) connectors for a 128128 switch system with 1128 planar lightwave circuit (PLC) type thermo-optic switches (TOSW). Input here the part of summary.

This paper proposes an adaptive noise canceller (ANC) with low signal-distortion for human-robot communication. The proposed ANC has two sets of adaptive filters for noise and crosstalk; namely, main filters (MFs) and subfilters (SFs) connected in parallel thereto. To reduce signal-distortion in the output, the stepsizes for coefficient adaptation in the MFs are controlled according to estimated signal-to-noise ratios (SNRs) of the input signals. This SNR estimation is carried out using SF output signals. The stepsizes in the SFs are determined based on the ratio of the primary and the reference input signals to cope with a wider range of SNRs. This ratio is used as a rough estimate of the input signal SNR at the primary input. Computer simulation results using TV sound and human voice recorded in a carpeted room show that the proposed ANC reduces both residual noise and signal-distortion by as much as 20 dB compared to the conventional ANC. Evaluation in speech recognition with this ANC reveals that with a realistic TV sound level, as good recognition rate as in the noise-free condition is achieved.

R-trees have been traditionally optimized for I/O performance with disk pages as tree nodes. Recently, researchers have proposed cache-conscious variations of R-trees optimized for CPU cache performance in main memory environments, where the node size is several cache lines wide and more entries are packed in a node by compressing MBR keys. However, because there is a big difference between the node sizes of two types of R-trees, disk-optimized R-trees show poor cache performance while cache-optimized R-trees exhibit poor disk performance. In this paper, we propose a cache and disk optimized R-tree, called PR-tree (Prefetching R-tree). For cache performance, the node size of the PR-tree is wider than a cache line, and the prefetch instruction is used to reduce the number of cache misses. For I/O performance, the nodes of the PR-tree are fitted into one disk page. We represent the detailed analysis of cache misses for range queries, and enumerate all the reasonable in-page leaf and nonleaf node sizes, and heights of in-page trees to figure out tree parameters for the best cache and I/O performance. The PR-tree that we propose achieves better cache performance than the disk-optimized R-tree: a factor of 3.5-15.1 improvement for one-by-one insertions, 6.5-15.1 improvement for deletions, 1.3-1.9 improvement for range queries, and 2.7-9.7 improvement for k-nearest neighbor queries. All experimental results do not show notable declines of I/O performance.

The motion of arc spots of breaking arc is investigated for Ag electrical contacts in DC 42 V/10 A resistive circuit using a high-speed camera. Also, the eroded contact surfaces are observed with a microscope after each breaking operation. As results, some kinds of different films and eroded regions are distinguished. Diameters of these regions are corresponding to the widths of the cathode and anode spot regions that are obtained by using the high-speed camera. It is found that the films and eroded regions on the electrical contacts are generated at different stages of the breaking arc.

In each contact material (Ag, Cu, Ni, and Fe), the breaking arcs occurring between an electrical contact pair in a resistive circuit of DC42 V/10.5 A were observed with a high-speed camera (1000 frames/s). Arc voltage and arc current were also measured simultaneously. By analyzing cathode and anode bright spots in the photographs, the positions of cathode and anode bright spots on contact surfaces were plotted on the graph. As a result, cathode and anode bright spots were found to express the characteristic motion in each material. Moreover, by comparing those results with the photograph of contact surface after all operations.

When a mobile node that subscribes to one or more multicast groups moves to another subnet, it is essential to provide a network level multicast handoff mechanism. Previous multicast handoff schemes are based on Mobile IP. However it is known that the Mobile IP is not adequate to interactive multimedia applications such as voice over IP or video conferencing due to its large handoff delay. Additionally, few researches have paid attentions on multicast handoff in infrastructure-mode WLAN environment. This paper proposes a fast inter-subnet multicast handoff method in Mobile IP based infrastructure-mode IEEE 802.11 WLAN environment. We introduce a dedicated Multicast Access Point (MAP) that works with an access points specified in standard IEEE 802.11 WLAN in order to alleviate disruption of receiving multicast datagram. Unlike previous research, our scheme does not modify Mobile IP specifications. MAP detects the completion of link-layer handoff, sends unsolicited IGMP Membership report to its local router on behalf of the mobile station and performs unicast tunneling. We evaluate the proposed method using ns-2 simulation. The simulation result shows that the proposed method can reduce the disruption period due to inter-subnet multicast handoff to about 1/12 and the packet loss rate can be reduced to about 1/4 over 20-size multicast group compared with the standard Mobile IP based IEEE 802.11 WLAN.

The application of neural networks to the state-feedback guaranteed cost control problem of discrete-time system that has uncertainty in both state and input matrices is investigated. Based on the Linear Matrix Inequality (LMI) design, a class of a state feedback controller is newly established, and sufficient conditions for the existence of guaranteed cost controller are derived. The novel contribution is that the neurocontroller is substituted for the additive gain perturbations. It is newly shown that although the neurocontroller is included in the discrete-time uncertain system, the robust stability for the closed-loop system and the reduction of the cost are attained.

Modern telecom and signal relays have been optimized to carry and switch low signals and to withstand high dielectric strength. Recent designs have extremely small physical dimensions and are comparatively cheap. Small size and low cost also make this type of relay very attractive for industrial and automotive applications. For industrial and automotive applications performance characteristics other than low and stable contact resistance values are of importance. While, for industrial applications, safety aspects and inductive load switching characteristics are of major importance, in automotive applications, high switching currents, inductive and lamp loads and high ambient temperatures are essential. Tests were carried out in order to determine the limitations of small size relays. The results obtained clearly show the unexpectedly high load range which signal relays are able to cover. Despite their small size, these relays can handle switching loads up to several hundred volts and currents up to 5 A. On top of the high switching current there is high excess current capability, and relays can work at extreme ambient temperatures between -55 and more than +105 degrees C.

This paper presents a study on the blind separation of a convoluted mixture of speech signals using Frequency Domain Independent Component Analysis (FDICA) algorithm based on the negentropy maximization of Time Frequency Series of Speech (TFSS). The comparative studies on the negentropy approximation of TFSS using generalized Higher Order Statistics (HOS) of different nonquadratic, nonlinear functions are presented. A new nonlinear function based on the statistical modeling of TFSS by exponential power functions has also been proposed. The estimation of standard error and bias, obtained using the sequential delete-one jackknifing method, in the approximation of negentropy of TFSS by different nonlinear functions along with their signal separation performance indicate the superlative power of the exponential-power-based nonlinear function. The proposed nonlinear function has been found to speed-up convergence with slight improvement in the separation quality under reverberant conditions.

This paper presents a design method for three-level programmable logic arrays (PLAs), which have input decoders and two-input EXOR gates at the outputs. The PLA realizes an EXOR of two sum-of-products expressions (EX-SOP) for multiple-valued input two-valued output functions. We developed an output phase optimization method for EX-SOPs where some outputs of the function are minimized in the complemented form and presented techniques to minimize EX-SOPs for adders by using an extension of Dubrova-Miller-Muzio's AOXMIN algorithm. The proposed algorithm produces solutions with a half products of AOXMIN-like algorithm in 250 times shorter time for large adders with two-valued inputs. We also proved that an n-bit adder with two-valued inputs requires at most 32n-2+7n-5 products in an EX-SOP while it is known that a sum-of-products expression (SOP) requires 62n-4n-5 products.

In this paper, we propose a statistical model for detecting article errors, which Japanese learners of English often make in English writing. It is based on the three head words--the verb head, the preposition, and the noun head. To overcome the data sparseness problem, we apply the backed-off estimate to it. Experiments show that its performance (F-measure=0.70) is better than that of other methods. Apart from the performance, it has two advantages: (i) Rules for detecting article errors are automatically generated as conditional probabilities once a corpus is given; (ii) Its recall and precision rates are adjustable.

Ad hoc networks have recently become a hot topic. In ad hoc networks, battery power is an important resource, since most terminals are battery powered. Terminals consume extra energy when their network interfaces are in the idle state or when they overhear packets not destined for them. They should, therefore, switch off their radio when they do not have to send or receive packets. IEEE802.11 features a power saving mechanism (PSM) in Distributed Coordination Function(DCF). In PSM for DCF, nodes must stay awake for a fixed time, called ATIM window (Ad-Hoc Traffic Indication Map window). If nodes do not have data to send or receive, they enter the doze state except for during ATIM window. However, ad hoc networks with PSM have longer end-to-end delays to deliver packets and suffer lower throughput than the standard IEEE802.11. To solve this problem, this paper proposes a protocol that reduces delay and achieves high throughput and energy efficiency. Simulation results show that our proposal outperforms other PSMs in terms of throughput, end-to-end delay and energy efficiency.

This paper focuses on the underdetermined blind source separation (BSS) of three speech signals mixed in a real environment from measurements provided by two sensors. To date, solutions to the underdetermined BSS problem have mainly been based on the assumption that the speech signals are sufficiently sparse. They involve designing binary masks that extract signals at time-frequency points where only one signal was assumed to exist. The major issue encountered in previous work relates to the occurrence of distortion, which affects a separated signal with loud musical noise. To overcome this problem, we propose combining sparseness with the use of an estimated mixing matrix. First, we use a geometrical approach to detect when only one source is active and to perform a preliminary separation with a time-frequency mask. This information is then used to estimate the mixing matrix, which allows us to improve our separation. Experimental results show that this combination of time-frequency mask and mixing matrix estimation provides separated signals of better quality (less distortion, less musical noise) than those extracted without using the estimated mixing matrix in reverberant conditions where the reverberant time (TR) was 130 ms and 200 ms. Furthermore, informal listening tests clearly show that musical noise is deeply lowered by the proposed method comparatively to the classical approaches.

A robust microphone array for speech enhancement and noise suppression is studied in this paper. To overcome target signal cancellation problem of conventional beamformer caused by array imperfections or reverberation effects of acoustic enclosure, the proposed microphone array adopts an arbitrary model of channel transfer function (TF) relating microphone and speech source. Since the estimation of channel TF itself is often intractable, herein, transfer function ratio (TFR) is estimated instead and used to form a suboptimal beamformer. A robust TFR estimation method is proposed based on signal subspace analysis technique against stationary or slowly varying noise. Experiments using simulated signal and actual signal recorded in a real room illustrate that the proposed method has high performance in adverse environment.

A speech signal captured by a distant microphone is generally smeared by reverberation, which severely degrades both the speech intelligibility and Automatic Speech Recognition (ASR) performance. Previously, we proposed a single-microphone dereverberation method, named "Harmonicity based dEReverBeration (HERB)." HERB estimates the inverse filter for an unknown room transfer function by utilizing an essential feature of speech, namely harmonic structure. In previous studies, improvements in speech intelligibility was shown solely with spectrograms, and improvements in ASR performance were simply confirmed by matched condition acoustic model. In this paper, we undertook a further investigation of HERB's potential as regards to the above two factors. First, we examined speech intelligibility by means of objective indices. As a result, we found that HERB is capable of improving the speech intelligibility to approximately that of clean speech. Second, since HERB alone could not improve the ASR performance sufficiently, we further analyzed the HERB mechanism with a view to achieving further improvements. Taking the analysis results into account, we proposed an appropriate ASR configuration and conducted experiments. Experimental results confirmed that, if HERB is used with an ASR adaptation scheme such as MLLR and a multicondition acoustic model, it is very effective for improving ASR performance even in unknown severely reverberant environments.

This paper describes an estimation of the computational and memory complexities of Greengard-Rokhlin's Fast Multipole Algorithm (GRFMA). GRFMA takes a quad tree structure and six calculation processes. We consider a perfect a-ary tree structure and the number of floating-point operations for each calculation process. The estimation for both complexities shows that the perfect quad tree is the best and the perfect binary tree is the worst. When we apply GRFMA to the computation of realistic problems, volume scattering are the best case and surface scattering are the worst case. In the worst case, the computational and memory complexities of GRFMA are O(Llog2 L) and O(Llog L), respectively. The computational complexity of GRFMA is higher than that of the multilevel fast multipole algorithm.

This paper proposes a reconfigurable adaptive FEC system based on Reed-Solomon (RS) code with interleaving. In adaptive FEC schemes, error correction capability t is changed dynamically according to the communication channel condition. For given error correction capability t, we can implement an optimal RS decoder composed of minimum hardware units for each t. If the hardware units of the RS decoder can be reduced for any given error correction capability t, we can embed as large deinterleaver as possible into the RS decoder for each t. Reconfiguring the RS decoder embedded with the expanded deinterleaver dynamically for each error correction capability t allows us to decode larger interleaved codes which are more robust error correction codes to burst errors. In a reliable transport protocol, experimental results show that our system achieves up to 65% lower packet error rate and 5.9% higher data transmission throughput compared to the adaptive FEC scheme on a conventional fixed hardware system. In an unreliable transport protocol, our system achieves up to 76% better bit error performance with higher code rate compared to the adaptive FEC scheme on a conventional fixed hardware system.

Recently, automatic speech recognition in a car has practical uses for applications like car-navigation and hands-free telephone dialers. For noise robustness, the current successes are based on the assumption that there is only a stationary cruising noise. Therefore, the recognition rate is greatly reduced when there is music or news coming from a radio or a CD player in the car. Since reference signals are available from such in-vehicle units, there is great hope that echo cancellers can eliminate the echo component in the observed noisy signals. However, previous research reported that the performance of an echo canceller is degraded in very noisy conditions. This implies it is desirable to combine the processes of echo cancellation and noise reduction. In this paper, we propose a system that uses echo cancellation and spectral subtraction simultaneously. A stationary noise component for spectral subtraction is estimated through the adaptation of an echo canceller. In our experiments, this system significantly reduced the errors in automatic speech recognition compared with the conventional combination of echo cancellation and spectral subtraction.