We propose a new band selective stop filter construction to decrease the out of band intermodulation distortion (IMD) noise generated in the transmitting power amplifier. Suppression of IMD noise directly improves the adjacent channel leakage power ratio (ACLR). A high-temperature superconducting (HTS) device with extremely high-Q performance with very small hybrid IC pattern would make it possible to implement the proposed filter construction as a practical device. To confirm the effectiveness of the HTS reaction-type filter (HTS-RTF) in improving ACLR, investigations based on both experiments and numerical analyses are carried out. The structure of a 5-GHz split open-ring resonator is investigated; its targets include high-unload Q-factor, low current densities, and low radiation. A designed 5-GHz HTS-RTF with 4 MHz suppression bandwidth and more than 40 dB MHz-1 sharp skirt is fabricated and experimentally investigated. The measured ACLR values are improved by a maximum of 12.8 dB and are constant up to the passband signal power of 40 dBm. In addition, to examine the power efficiency improvement offered by noise suppression of the HTS-RTF, numerical analyses based on measured results of gallium nitride HEMT power amplifier characteristics are conducted. The analyzed results shows the drain efficiency of the amplifier can be improved to 44.2% of the amplifier with the filter from the 15.7% of the without filter.

A method is proposed for estimating code and carrier phase parameters of GNSS reflected signals in low SNR (signal-to-noise ratio) environments. Simulation results show that the multipath impact on code and carrier with 0.022 C/A chips delay can be estimated in 0 dB SNR in the condition of 46 MHz sampling rate.

In this letter, a closed form approximation for the average sum rate of random beamforming is derived. It provides a near-exact approximation for arbitrary numbers of beams, users, and received SNR. The approximation is also applied to an average-sense multimode random beamforming scheme which optimizes the number of random beams without any type of instantaneous channel information. The proposed scheme shows better sum rate performance than random beamforming as well as an existing dual mode random beamforming scheme based on instantaneous channel information, while the number of feedback bits for beam index is reduced compared to random beamforming.

In this paper, a novel design method of critically sampled contourlet transform (CSCT) is proposed. The original CT which consists of Laplacian pyramid and directional filter bank provides efficient frequency plane partition for image representation. However its overcompleteness is not suitable for some applications such as image coding, its critical sampling version has been studied recently. Although several types of the CSCT have been proposed, they have problems on their realization or unnatural frequency plane partition which is different from the original CT. In contrast to the way in conventional design methods based on a "top-down" approach, the proposed method is based on a "bottom-up" one. That is, the proposed CSCT decomposes the frequency plane into small directional subbands, and then synthesizes them up to a target frequency plane partition, while the conventional ones decompose into it directly. By this way, the proposed CSCT can design an efficient frequency division which is the same as the original CT for image representation can be realized. In this paper, its effectiveness is verified by non-linear approximation simulation.

Two discrete-time Wirtinger-type inequalities relating the power of a finite-length signal to that of its circularly-convolved version are developed. The usual boundary conditions that accompany the existing Wirtinger-type inequalities are relaxed in the proposed inequalities and the equalizing sinusoidal signal is free to have an arbitrary phase angle. A measure of this sinusoidal signal's power, when corrupted with additive noise, is proposed. The application of the proposed measure, calculated as a ratio, in the evaluation of the power of a sinusoid of arbitrary phase with the angular frequency π/N, where N is the signal length, is thoroughly studied and analyzed under additive noise of arbitrary statistical characteristic. The ratio can be used to gauge the power of sinusoids of frequency π/N with a small amount of computation by referring to a ratio-versus-SNR curve and using it to make an estimation of the noise-corrupted sinusoid's SNR. The case of additive white noise is also analyzed. A sample permutation scheme followed by sign modulation is proposed for enlarging the class of target sinusoids to those with frequencies M π/N, where M and N are mutually prime positive integers. Tandem application of the proposed scheme and ratio offers a simple method to gauge the power of sinusoids buried in noise. The generalization of the inequalities to convolution kernels of higher orders as well as the simplification of the proposed inequalities have also been studied.

The IEEE802.11b WLAN standard employs direct-sequence/spread-spectrum (DS/SS) modulation. With a fractional sampling RAKE receiver, it is possible to achieve diversity and reduce the BER in DS/SS communication. In order to realize the diversity through fractional sampling, the impulse response of the channel must be estimated. In this paper, a channel estimation scheme for a RAKE receiver with fractional sampling in IEEE802.11b WLAN system is investigated through a computer simulation and an experiment. In order to estimate the impulse response of the channel, a pseudo-inverse matrix with a threshold is employed. Numerical results indicate that the channel can be estimated with an optimum threshold in both the simulation and the experiment.

A novel closed-form algorithm is presented for estimating the 3-D location (azimuth angle, elevation angle, and range) of a single source in a uniform circular array (UCA) with a center sensor. Based on the centrosymmetry of the UCA and noncircularity of the source, the proposed algorithm decouples and estimates the 2-D direction of arrival (DOA), i.e. azimuth and elevation angles, and then estimates the range of the source. Notwithstanding a low computational complexity, the proposed algorithm provides an estimation performance close to that of the benchmark estimator 3-D MUSIC.

Transmission delay in audio communications is a well-known obstacle to achieving smooth communication. However, it is not known what kinds of effects are caused by small delays. We hypothesized that the small delay in the listener's responses disturbs the speaker's "verbal conditioning," where the verbal behavior of the speaker varies in accordance with the listener's responses. We examined whether the small delays in the listener's responses disturb the speaker's verbal conditioning using an artificial-grammar learning task. The results suggested that a 300-ms delay disturbed the participants' verbal conditioning although they were not adequately aware of the delay.

In this paper, we propose a cross-layer design of packet level cooperation for wireless LANs that support rate adaptation. While keeping compatibility with legacy wireless LANs, distributed control of multi-hop packet transmission is enabled without pre-negotiation of routing or pairing. These features are provided by prioritization scheme based on IEEE 802.11e EDCF in which we set the parameters according to the measured link condition at each terminal. Relaying packets with high transmission rate makes much efficient use of radio resource, and it leads not only to improve performance of the total system, but also to overcome the fairness issue known in rate adaptation, where the terminals with good link conditions cannot gain enough resource because of the time consumed by low data rate transmission. These advantages are confirmed through computer simulations considering packet error rate at each transmission which is assumed by receiving power calculated from the distance between source, relay, and destination terminals. Furthermore, we also discuss about the fairness between cooperation-enabled and legacy terminals when they coexist in the same system, in order to make gradual deployment feasible.

An impedance-isolation technique is proposed for on-chip ESD protection design for radio-frequency (RF) integrated circuits (ICs), which has been successfully verified in a 0.25-µm CMOS process with thick top-layer metal. With the resonance of LC-tank at the operating frequency of the RF circuit, the impedance (especially, the parasitic capacitance) of the ESD protection devices can be isolated from the RF input node of low-noise amplifier (LNA). Therefore, the LNA can be co-designed with the proposed impedance-isolation technique to simultaneously achieve excellent RF performance and high ESD robustness. The power gain (S21-parameter) and noise figure of the ESD protection circuits with the proposed impedance-isolation technique have been experimentally measured and compared to those with the conventional double-diodes ESD protection scheme. The proposed impedance-isolation technique had been demonstrated to be suitable for on-chip ESD protection design for RF ICs.

The superconducting band-pass filter has small insertion loss and excellent out-of-band rejection properties. It has been put to practical use in a number of applications. However, in order to expand its range of application, a tuning technique that can restore the filter characteristics is needed. We propose an automatic tuning system using a trimming library and checked the feasibility of the system by tuning a forward-coupled filter with three resonators. The results show that the trimming library method is an effective way of automatically improving the filter characteristics.

A desired property of large distributed systems is self adaptability against the faults that occur more frequently as the size of the distributed system grows. Self-stabilizing protocols provide autonomous recovery from finite number of transient faults. Fault-containing self-stabilizing protocols promise not only self-stabilization but also containment of faults (quick recovery and small effect) against small number of faults. However, existing composition techniques for self-stabilizing protocols (e.g. fair composition) cannot preserve the fault-containment property when composing fault-containing self-stabilizing protocols. In this paper, we present Recovery Waiting Fault-containing Composition (RWFC) framework that provides a composition of multiple fault-containing self-stabilizing protocols while preserving the fault-containment property of the source protocols.

MPLS-based path technology shows promise as a means of realizing reliable IP networks. Real-time services such as VoIP and video-conference supplied through a multi-domain MPLS network must be able to guarantee end-to-end QoS of the inter-domain paths. Thus, it is important to allocate an appropriate QoS class to the inter-domain paths in each domain traversed by the inter-domain paths. Because each domain has its own policy for QoS class allocation, it is necessary to adaptively allocate the optimum QoS class based on estimation of the QoS class allocation policies in other domains. This paper proposes two kinds of adaptive QoS class allocation schemes, assuming that the arriving inter-domain path requests include the number of downstream domains traversed by the inter-domain paths and the remaining QoS value toward the destination nodes. First, a measurement-based scheme, based on measurement of the loss rates of inter-domain paths in the downstream domains, is proposed. This scheme estimates the QoS class allocation policies in the downstream domains, using the measured loss rates of path requests. Second, a state-dependent type scheme, based on measurement of the arrival rates of path requests in addition to the loss rates of paths in the downstream domains, is also proposed. This scheme allows an appropriate QoS class to be allocated according to the domain state. This paper proposes an application of the Markov decision theory to the modeling of state-dependent type scheme. The performances of the proposed schemes are evaluated and compared with those of the other less complicated non-adaptive schemes using a computer simulation. The results of the comparison reveal that the proposed schemes can adaptively increase the number of inter-domain paths accommodated in the considered domain, even when the QoS class allocation policies change in the other domains and the arrival pattern of path requests varies in the considered domain.

We address the issue of MIMO channel estimation with the aid of a priori temporal correlation statistics of the channel as well as the spatial correlation. The temporal correlations are incorporated to the estimation scheme by assuming the Gauss-Markov channel model. Under the MMSE criteria, the Kalman filter performs an iterative optimal estimation. To take advantage of the enhanced estimation capability, we focus on the problem of channel estimation from a partial channel measurement in the MIMO antenna selection system. We discuss the optimal training sequence design, and also the optimal antenna subset selection for channel measurement based on the statistics. In a highly correlated channel, the estimation works even when the measurements from some antenna elements are omitted at each fading block.

As a candidate for the transmission technology of next generation mobile communication systems, time-domain spreading MC-CDMA systems have begun to attract much attention. In these systems, data and pilot symbols are spread in the time domain and code-multiplexed. To combat fading issues, we need to conduct channel estimation by using the code-multiplexed pilot symbols. Especially in next generation systems, frequency bands higher than those of current systems, which raise the maximum Doppler frequency, are expected to be used, so that a more powerful channel estimation method is expected. Considering this, we propose a channel estimation method for highly accurate channel estimation; it is a combination of a two-dimensional channel estimation method and an impulse response-based channel estimation method. We evaluate the proposed method by computer simulations.

In this paper, we propose a subcarrier resource allocation algorithm for managing the video quality degradation for multiuser orthogonal frequency division multiplex (OFDM) systems. The proposed algorithm exploits the unequal importance existing in different picture types for video coding and the diversity of subcarriers for multiuser systems. A model-based performance metric is first derived considering the error concealment and error propagation properties of the H.264 video coding structure. Based on the information on video quality enhancement existing in a packet to be transmitted, we propose the distortion management algorithm for balancing the subcarriers and power usages for each user and minimizing the overall video quality degradation. In the simulation results, the proposed algorithm demonstrates a more gradual video quality degradation for different numbers of users compared with other resource allocation schemes.

Although software-attack detection via dynamic taint analysis (DTA) supports high coverage of program execution, it prohibitively degrades the performance of the monitored program. This letter explores the possibility of collaborative dynamic taint analysis among members of an application community (AC): instead of full monitoring for every request at every instance of the AC, each member uses DTA for some fraction of the incoming requests, thereby loosening the burden of heavyweight monitoring. Our experimental results using a test AC based on the Apache web server show that speedy detection of worm outbreaks is feasible with application communities of medium size (i.e., 250-500).

This paper shows that there is a fruitful world behind sampling theorems. For this purpose, the sampling problem is reformulated from a functional analytic standpoint, and is consequently revealed that the sampling problem is a kind of inverse problem. The sampling problem covers, for example, signal and image restoration including super resolution, image reconstruction from projections such as CT scanners in hospitals, and supervised learning such as learning in artificial neural networks. An optimal reconstruction operator is also given, providing the best approximation to an individual original signal without our knowing the original signal.

A method for measuring similarity between two variables is presented. Our approach considers the case where available observations are arbitrarily filtered versions of the variables. In order to measure the similarity between the original variables from the observations, we propose an error-minimizing filter (EMF). The EMF is designed so that an error between outputs of the EMF is minimized. In this paper, the EMF is constructed by a finite impulse response (FIR) filter, and the error between the outputs is evaluated by the mean square error (EMF). We show that minimization of the MSE results in an eigenvalue problem, and the optimal solution is given in a closed form. We also reveal that the minimal MSE by the EMF is efficient in the measurement of the similarity from the viewpoint of a correlation coefficient between the originals.

By use of the shadow theory developed recently, this paper deals with the transverse electric (TE) wave diffraction by a perfectly conductive periodic array of rectangular grooves. A set of equations for scattering factors and mode factors are derived and solved numerically. In terms of the scattering factors, diffraction amplitudes and diffraction efficiencies are calculated and shown in figures. It is demonstrated that diffraction efficiencies become discontinuous at an incident wave number where the incident wave is switched from a propagating wave to an evanescent one, whereas scattering factors and diffraction amplitudes are continuous even at such an incident wave number.