In this letter, we present a space division multiple access (SDMA) approach for IEEE802.11a-based system employing pre-fast Fourier transform (FFT) adaptive array antenna (AAA) at base station (BS). As the core idea, we propose a preamble subcarrier assignment method to generate different preambles for different users using the same signal burst structure defined by IEEE802.11a, by which BS can effectively distinguish each user from other users and accurately estimate the channel impulse response (CIR) for each user. In this way, SDMA can be easily realized with no significant change in IEEE802.11a-based system. The performance of the proposed SDMA system is evaluated by computer simulation using a realistic spatio-temporal indoor wireless channel model.

In label recognition based on optical correlation processing, to completely discriminate a 4-bit target optical code from all types of 4-bit nontarget ones, we propose a novel label recognition method using both optical time-gating and the designed label recognition filter. We experimentally demonstrate that the intensities of correlation signals of 4-bit similar optical codes can be suppressed by the designed label recognition filter and that only the correlation signal of a 4-bit target optical code can be detected by extraction with optical time-gating. The optical time-gating is realized by using organic nonlinear optical crystal: 2-adamantylamino-5-nitropyridine (AANP).

The wireless streaming media communications are fragile to the delay jitter because the conditions and requirements vary frequently with the users' mobility. Buffering is a typical way to reduce the delay jitter of media packets before the playback, however, it will incur a longer end-to-end delay. Our motivation in this paper is to improve the balance between the elimination of delay jitter and the decrease of end-to-end delay. We propose a novel adaptive playback buffer (APB) based on the probing scheme. By utilizing the probing scheme, the instantaneous network situations are collected, and then the delay margin and the delay jitter margin are employed to calculate the step length (sl) which is used to adjust the playback buffer in each time. The adaptive adjustment to the playback buffer in APB enables the continuous and real-time representation of streaming media at the receiver. Unlike the previous studies, the novelty and contributions of the paper are: a) Accuracy: by employing the instantaneous network information, the adjustment to the playback buffer correctly reflects the current network situations and therefore achieves the improved balance between the elimination of delay jitter and the decrease of end-to-end delay; Hence, APB adjustment is accurate in terms of improving such balance; b) Efficiency: by utilizing the simple probing scheme, APB achieves the current network situations without the complex mathematic predictions, which enables the adjustment to be more timely and efficient. Performance data obtained through extensive simulations show that our APB is effective to reduce both delay jitter and playback buffer delay.

This paper is devoted to the problem of force sensorless disturbance rejection in robot manipulators. In the proposed approach, the control system uses position sensor signals and estimated values of external forces, instead of force sensor signals. The estimation process is performed via an adaptive force estimator. Then the estimated force vector is utilized to compensate for the force disturbance effect in order to achieve a better trajectory tracking performance. The force estimation is carried out directly using no environment model. Asymptotical stability of the proposed control system is analyzed by the invariant set and Lyapunov direct method establishing an appropriate theorem. Finally, the performance of the proposed control system is verified using numerical simulation.

This paper introduces a state-of-art on an ultra-wideband (UWB) technology in intelligent transport systems (ITS). To examine the detection performance of a UWB short-range radar for vehicular applications, we developed a 26-GHz band short-range UWB radar system with an embedded compact MMIC-based RF module. In this paper, we briefly comment on the current regulatory environment for UWB radar systems by outlining the structure of an international organization involved in examining the regulatory status of these systems. We then describe the principles of detection and system design for impulse radar, the radar system that we developed, and a MMIC-based RF module as well as the performance of these devices. We measured their performance in a series of laboratory experiments and also measured UWB radar cross sections of an automobile. The results of our experiments suggest that our radar system is capable of detecting targets with a range resolution of around 9 cm.

The purpose of this paper is to improve a feedback-type adaptive array antenna (AAA) with feedback information quantized by one bit which was presented recently on TDMA system by an author of this paper. The improvement is made by using adaptive, instead of constant, update size of adaptive antenna weights control. Computer simulation results show that the performance of this system is improved to be almost equivalent to the performance of a system without quantization of the feedback information for wide range of fading speed. The results include the effect of control delay time and the maximum Doppler frequency under flat fading and frequency-selective fading.

In contention-based wireless ad hoc networks, power control is an efficient way to improve the spatial reuse by allowing multiple pairs to communicate simultaneously. In this paper, we propose a game-theoretical approach for joint power and rate control in ad hoc networks, where the transmit rate of each link is maximized. Meanwhile we consider the transmit power as the cost, since higher power leads to higher interference and more energy consumption. In particular, we introduce a novel auction-like pricing algorithm in which the cost per unit power steps up until the network settles down at a Nash equilibrium, which is a feasible power and rate allocation, even when the Signal to Interference and Noise Ratio (SINR) requirements are initially infeasible. Numerical results show significant throughput improvement and energy consumption savings compared with the previously proposed algorithm that defers the link with minimum SINR.

We investigate the optimal chip rate of power or rate adapted direct-sequence code division multiple access (DS/CDMA) communication systems in Nakagami fading channels. We find that the optimal chip rate that maximizes the spectral efficiency depends upon both the channel parameters, such as multipath intensity profile (MIP) and line-of-sight (LOS) component, and the adaptation scheme itself. With the rate adaptation, the optimal chip rate is less than 1/Tm irrespective of the channel parameters, where Tm is multipath delay spread. This indicates that with the rate adaptation, correlation receiver achieves higher spectral efficiency than RAKE receiver. With the power adaptation, however, the optimal chip rate and the corresponding number of tabs in RAKE receiver are sensitive to MIP and LOS component.

In this paper, we present a novel iterative MPEG super-resolution method based on an embedded constraint version of Adaptive projected subgradient method [Yamada & Ogura 2003]. We propose an efficient operator that approximates convex projection onto a set characterizing framewise quantization, whereas a conventional method can only handle a convex projection defined for each DCT coefficient of a frame. By using the operator, the proposed method generates a sequence that efficiently approaches to a solution of super-resolution problem defined in terms of quantization error of MPEG compression.

This paper proposes the application of tree-structured clustering to the processing of noisy speech collected under various SNR conditions in the framework of piecewise-linear transformation (PLT)-based HMM adaptation for noisy speech. Three kinds of clustering methods are described: a one-step clustering method that integrates noise and SNR conditions and two two-step clustering methods that construct trees for each SNR condition. According to the clustering results, a noisy speech HMM is made for each node of the tree structure. Based on the likelihood maximization criterion, the HMM that best matches the input speech is selected by tracing the tree from top to bottom, and the selected HMM is further adapted by linear transformation. The proposed methods are evaluated by applying them to a Japanese dialogue recognition system. The results confirm that the proposed methods are effective in recognizing digitally noise-added speech and actual noisy speech issued by a wide range of speakers under various noise conditions. The results also indicate that the one-step clustering method gives better performance than the two-step clustering methods.

Ultra Wideband (UWB) communication system utilizing impulse signals is attractive technique which can achieve high data rate with low complexity and low power consumption. In this impulse based UWB system, lots of different shaped pulses have been considered to represent more information bits per symbol. In order to detect these different shaped UWB signals at the receiver, the synthesized template generation method using several elementary waveforms is effective. In this paper we design and analyze this synthesized template waveform instead of the conventional matched filter technique. The synthesis of UWB template waveform can be achieved as combinations of orthogonalized elementary waveforms with Fourier coefficients. By adjusting the number of elementary waveforms and their coefficients, it is possible to detect several types of UWB signals. The orders of approximation corresponding to different number of elementary waveforms are analyzed and the bit error rate properties are then investigated in AWGN and multipath fading channels. In addition, the proposed system can capture more energy by adjusting its coefficients adaptively under the multipath environment and reduce the effect of Intra-Pulse Interference (IPI) which is occurred when the propagation channel is not separable, that is, multipath components spaced closer than the typical pulse width. We show the design of the adaptive template synthesis method and its performance compared with conventional Rake receiver.

Adaptive Virtual Queue (AVQ) introduces a novel implementation algorithm for Active Queue Management (AQM). The stability criterion for AVQ was deduced in literature [1], but it lacks practicability due to the difficulty of solving the transcendental equation. In this letter, the AVQ stability is further investigated based on the characteristic roots of delay-differential equation. Another stability criterion explicitly associated with parameters of network configuration is deduced and the upper bound of delay time for stable AVQ algorithm is determined. Finally, the conclusion is validated through simulation experiments.

The transient scattering characteristics of millimeter waves from a cylindrical object near a flat boundary were measured by the 50 GHz scatterometer to evaluate the multiple interactions of scattered waves with the objects and the boundary. Both perfectly conducting and dielectric cylinders are considered as a scattering object. The pulse intensities including waves scattered first from the object and then from the flat boundary or vice versa are shown to be significantly influenced by the distance from the object to the boundary, depending on the refractive index of the object. The observed higher order responses including the multiple scattering between the object and the boundary are also discussed. A preliminary comparison of the measured and calculated pulse responses for the perfectly conducting object is presented at slightly oblique incidence on a flat boundary.

The present paper discusses a new construction of UWB pulses within the framework of soft-spectrum adaptation. The employed basis functions are B-splines having the following properties: (i) The B-splines are time-limited piecewise polynomials. (ii) The first-order B-splines are rectangular pulses and they converge band-limited functions at the limit that their order tends to infinity. (iii) There are an analog circuit and a fast digital filter for the generation of B-splines. Simple application of Gram-Schmidt orthonormalization process to the shifted B-splines results in a few basic pulses, which are well time-limited and have a broad band width, but do not comply with the FCC spectral mask. A constrained approximation technique is proposed for adaptively designing pulses so that they approximate target frequency characteristics. At the cost of using eleven shifted B-splines, an example set of four pulses comforting the FCC spectral mask is obtained.

We investigate the impact of symbol rate control, modulation level control, and the number of hops on the area spectral efficiency of interference-limited multihop radio networks. By controlling symbol rate and modulation level, data rate can be adapted according to received power. In addition, varying the number of hops can control received power. First, we evaluate the achievable end-to-end throughput of multihop transmission assuming symbol rate and modulation level control. Numerical results reveal that by controlling symbol rate or using multihop transmission, the end-to-end communication range can be extended at the cost of end-to-end throughput, and this may result in lower area spectral efficiency. Next, an expression for the area spectral efficiency of multihop radio networks is derived as a function of the number of hops and the end-to-end throughput. Numerical results also reveal that the resulting area spectral efficiency depends on the specific circumstances, which, however, can be increased only by using multihop transmission.

Since WLAN (wireless LAN) systems share the 2.4-GHz frequency band with microwave ovens, interference caused by radiated oven noise is a serious problem in practical WLAN application. To mitigate the oven noise interference in DS-SS (direct-sequence spread spectrum) WLAN systems, the use of adaptive filters is proposed. This method is based on the fact that oven noise behaves like CW (continuous wave) interference within a short duration. In contrast to previous reduction techniques for oven noise, this method can be implemented without changing any specifications of current WLAN systems. The results of numerical and experimental analyses clearly demonstrate the effectiveness of adaptive filters for improving the bit error rates of WLAN links subject to oven noise interference.

Direction-of-arrival (DOA) estimation based on subspace methods has collected much interest over a few decades, and adaptive DOA estimation with rapidly changing parameters will be necessary for wireless communications. This paper is concerned with a new subspace tracking scheme by using an accelerated LMS and RLS algorithms for time-varying parameters. The proposed accelerated adaptive algorithms are based on the internal model principle by approximately expressing the changing parameters by an expansion of polynomial time functions. Thus its application to DOA estimation based on the MUSIC and MODE schemes is presented and the effectiveness is validated in numerical simulations.

The emergence of intelligent and sophisticated attack techniques makes web services more vulnerable than ever which are becoming an important business tool in e-commerce. Many techniques have been proposed to remove the security vulnerabilities, yet have limitations. This paper proposes an adaptive mechanism for a web-server intrusion-tolerant system (WITS) to prevent unknown patterns of attacks by adapting known attack patterns. SYN flooding attacks and their adaptive defense mechanisms are simulated as a case study to evaluate the performance of the proposed adaptation mechanism.

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.

This paper presents a theoretical convergence analysis of a CORDIC-based adaptive ARMA lattice filter. In previous literatures, several investigation methods for adaptive lattice filters have been proposed; however, they are available only for AR-type filters. Therefore, we have developed a distinct technique that can reveal the convergence properties of the CORDIC ARMA lattice filter. The derived technique provides a quantitative convergence analysis, which facilitates an efficient hardware design for the filter. Moreover, our analysis technique can be applied to popular multiplier-based filters by slight modifications. Hence, the presented convergence analysis is significant as a leading attempt to investigate ARMA lattice filters.