A number of schemes have been proposed for communication using chaos over the past years. Regardless of the exact modulation method used, the transmitted signal must go through a physical channel which undesirably introduces distortion to the signal and adds noise to it. The problem is particularly serious when coherent-based demodulation is used because the necessary process of chaos synchronization is difficult to implement in practice. This paper addresses the channel distortion problem and proposes a technique for channel equalization in chaos-based communication systems. The proposed equalization is realized by a modified recurrent neural network (RNN) incorporating a specific training (equalizing) algorithm. Computer simulations are used to demonstrate the performance of the proposed equalizer in chaos-based communication systems. The Henon map and Chua's circuit are used to generate chaotic signals. It is shown that the proposed RNN-based equalizer outperforms conventional equalizers.

In this letter, we investigate the blind channel estimation in MIMO-OFDM systems based on the second-order statistics of the channel outputs only. We exploit the cyclostationarity induced by OFDM with cyclic prefix, establish the sufficient identifiability conditions, and develop a subspace algorithm. Finally, we demonstrate the validity of the algorithm by computer simulations.

Let us introduce n ( 2) mappings fi (i=1,2,,n) defined on complete linear metric spaces (Xi-1, ρ) (i=1,2,,n), respectively, and let fi:Xi-1 Xi be completely continuous on bounded convex closed subsets Xi-1(0) Xi-1, (i=1,2,,n 0), such that fi(Xi-1(0)) Xi(0). Moreover, let us introduce n set-valued mappings Fi : Xi-1 Xi (Xi)(the family of all non-empty closed compact subsets of Xi), (i=1,2,,n 0). Here, we have a fixed point theorem on the successively recurrent system of set-valued mapping equations: xi Fi(xi-1, fi(xi-1)), (i=1,2,,n 0). This theorem can be applied immediately to analysis of the availability of system of circular networks of channels undergone by uncertain fluctuations and to evaluation of the tolerability of behaviors of those systems. In this paper, mathematical situation and detailed proof are discussed, about this theorem.

In loudspeaker-based 3D audio systems, there are some acoustic crosstalk cancellation methods to enlarge the 'sweet spot' around a fixed listener position. However, these methods have common defect that most of them can be applied only to the specific narrow frequency band. In this letter, we propose the more robust acoustic crosstalk cancellation method so that we can cancel the crosstalk signal in far wider frequency band and enlarge 'sweet spot. ' For this goal, we apply a sum and difference filter to the conventional three loudspeaker-based 3D audio system.

Pilot-aided adaptive prediction channel estimation is proposed for coherent detection in a frequency-nonselective fading channel. It is an extension of the conventional weighted multi-slot averaging (WMSA) channel estimation and consists of 3 steps. A block of Np pilot symbols is periodically transmitted, each pilot block being followed by Nd data symbols to form a data slot. In the first step, the instantaneous channel gain is estimated by coherent addition of Np pilot symbols. Using the K past and K future estimated instantaneous channel gains, the second step predicts the instantaneous channel gains at the end and beginning of data slot of interest by a forward predictor and a backward predictor, respectively. The tap-weights of forward prediction and backward prediction are adaptively updated using the normalized least mean square (NLMS) algorithm. Finally, in the third step, the instantaneous channel gain at each data symbol position within the data slot of interest is estimated by simple averaging or linear interpolation using the two adaptively predicted instantaneous channel gains. The computer simulation confirms that the proposed adaptive prediction channel estimation achieves better bit error rate (BER) performance than the conventional WMSA channel estimation in a fast fading channel and/or in the presence of frequency offset between a transmitter and a receiver.

In DS-CDMA (Direct Sequence-Code Division Multiple Access) mobile systems, soft handoff is recognized as reducing interference and increasing reverse link capacity, whereas from the radio channel resource viewpoint, soft handoff can make the system performance deteriorate significantly. This paper focuses on using the channel resources of soft handoff in a limited field environment for evaluating the performance of DS-CDMA mobile systems. Our traffic model is based on the mobile traffic environment with region overlap among multiple cells. The soft handoff rates are estimated by the mobility of mobiles, and simulation results show good agreement with the traffic model. The channel holding time distribution of the soft handoff traffic model, which does not have a closed-form solution, is derived by a numerical integration method. Computer simulations show promising results based on the non-closed-form application. We analyze the performance of DS-CDMA mobile systems with the proposed traffic model to see the effect of the soft handoff region on blocking probability. From the result, when the soft handoff region is extended with the given channels, both the handoff rate and the mean channel holding time are increased and these make the blocking probability increase a great deal. To maintain the required blocking probability as before, additional channel resources should be supplied according to the extended region. It is our belief that the proposed traffic model and the performance analysis presented in this paper are practically acceptable.

We describe a method of compensating temperature fluctuation by a linear-time-warping processing in a sound reproduction system. This technique is applied to impulse responses of room transfer functions, to achieve a high-quality sound reproduction system, particularly one that treats high-frequency components. First, the impulse responses are measured before and after temperature fluctuation, and the former are converted to the latter by the proposed process. Next, we design inverse filters for the system, and evaluate the improvement of the reproduction accuracy and spectrum distortion. By the compensation method, we can improve the reproduction accuracy at any frequency. Moreover, we propose an adaptive algorithm for the estimation of a suitable warping ratio, using the observed signal of reproduced sound obtained at only one control point. Using the proposed algorithm, we can improve the reproduction accuracy at each control point by about 14 dB, in which a difference in temperature is 1.4.

This paper proposes a pilot channel assisted minimum mean square error (MMSE) combining scheme in orthogonal frequency and code division multiplexing (OFCDM) based on actual signal-to-interference power ratio (SIR) estimation, and investigates the throughput performance in a broadband channel with a near 100-MHz bandwidth. In the proposed MMSE combining scheme, the combining weight of each sub-carrier component is accurately estimated from the channel gain, noise power, and transmission power ratio of all the code-multiplexed channels to the desired one, by exploiting the time-multiplexed common pilot channel in addition to the coded data channel. Simulation results elucidate that the required average received signal energy per bit-to-noise spectrum density ratio (Eb/N0) for the average packet error rate (PER) = 10-2 is improved by 0.6 and 1.2 dB by using the proposed MMSE combining instead of the conventional equal gain combining (EGC) in a 24-path Rayleigh fading channel (exponential decay path model, maximum delay time is approximately 1 µsec) in an isolated cell environment, when the number of multiplexed codes = 8 and 32, respectively, with the spreading factor of 32. Furthermore, when the average received Eb/N0 = 10 dB, the achievable throughput, i.e., the number of simultaneously multiplexed codes for the average PER = 10-2 in the proposed MMSE combining, is increased by approximately 1.3 fold that of the conventional EGC.

We propose using a soft decision-directed least-mean-square algorithm in a code-aided equalization scheme for fading channels. Soft-output Viterbi algorithm (SOVA) is modified and applied to strengthen the second-stage equalizer adaptation in the scheme. Simulation results are presented for bit error rate performance in a multipath environment for various normalized fade rates. The proposed equalizer scheme is shown to provide significant bit error rate improvement compared to conventional equalization schemes.

This paper discusses self-similarity in cell dwell time of a mobile terminal, the discovery of which was described in our previous paper, and its effects on teletraffic of mobile communication networks. We have evaluated various teletraffic statistics, such as cell dwell time and channel occupancy time, of a mobile terminal based on measurements of motion for various types of vehicles. Those results show that cell dwell time follows a long-tailed log-normal distribution rather than the exponential distribution that has been used for modeling. Here, we first elaborate on self-similarity in cell dwell time of various vehicles. We then evaluate self-similarity in channel occupancy time. For future mobile multimedia communication systems employing a micro-cell configuration, it is anticipated that data communication will be the main form of communication and that call holding time will be long. For such cases, we have shown that channel occupancy time will be greatly affected by the cell dwell time of the mobile terminal, and that self-similarity, a characteristic that is not seen in conventional systems, will consequently appear. We have also found that hand-off frequently fails as self-similarity in cell dwell time of a mobile terminal becomes stronger.

We consider bandwidth-efficient modulation schemes for use on non-linear channels, such as that due to a non-linear high power amplifier (HPA) in a wireless system. Continuous Phase Modulation (CPM) schemes are known to perform well on such channels, because they have constant amplitude, but their bandwidth efficiency is low. N-MSK improves this by superposing two or more such signals, but this results in a non-constant amplitude. In this paper we investigate the performance of N-MSK on a non-linear channel, modelled using a travelling wave tube (TWT) non-linearity. We first consider the spectrum and the BER performance of N-MSK on a non-linear channel, making use of the Euclidean distance spectrum of the modulated signal. We then consider the effect of non-linear amplification on these properties. Signal spectrum was determined by simulation, since no closed-form expression is available when the effect of AM-PM conversion are included. We find that the spectrum is remarkably little affected, being only slightly broadened. BER is also evaluated by direct simulation, as well as from the Euclidean distance spectrum. The latter now exhibits a series of clusters, instead of discrete lines, and we find that at least the whole of the first cluster must be considered in calculating the BER, not just the minimum distance or the centroid of the cluster. The detector used in the simulation applies an inverse distortion function, then uses maximum-likelihood sequence estimation (MLSE) set up for the linear channel. This is no longer optimum, because the noise is distorted, and therefore it is also compared with a true MLSE detector. We find that the BER performance is, however, somewhat degraded compared to the linear channel. We determine back-off levels from saturation to optimise overall power efficiency.

It has been shown that the performances of single-receiver ARQ schemes are largely dependent on the packet-error process, i.e., for dependent packet-error environment, they are under- or over- estimated by analyzing them under the assumption that packet-errors occur at random. While, multi-receiver ARQ's have not been analyzed and evaluated for dependent packet-error process. In this paper, we analyze the throughput efficiency of the fundamental multi-receiver Go-back-N ARQ scheme, which can be implemented very simply, over an unreliable channel modeled by the two-state Markov process. Any receiver erroneously receives a packet with probability inherent to each state. From numerical results, we show that the throughput efficiency of the fundamental multi-receiver Go-Back-N ARQ scheme depends on the number of receivers, round-trip-delay, and the characteristic of the Markov process. Also we show that the throughput efficiency of the fundamental multi-receiver Go-Back-N ARQ scheme for larger decay factor and larger difference between packet error probability at each state is considerably better than that for the random error pattern.

In this paper, we propose an efficient dual-tree-based multicasting scheme with three destination-switch partition strategies on irregular switch-based networks. The dual-tree-based routing scheme supports adaptive, distributed, and deadlock-free multicast on irregular networks with double channels. We first describe a dual-tree structure constructed from the irregular networks and prove that the multicasting based on such a structure is deadlock-free. Then, an efficient multicast routing algorithm with three destination-switch partition strategies: source-switch-based partition, destination-switch-based partition, and all-switches-based partition, is proposed. Finally, we perform simulations to evaluate our proposed algorithm under various impact parameters: system size, message length, and startup time. The experimental results show that the improved tree-based multicasting scheme outperforms the usual tree-based multicasting scheme. The dual-tree-based multicasting scheme with destination-switch-based partition is shown to be the best for all situations.

Blind adaptive channel identification of communication channels is a problem of important current theoretical and practical concerns. Recently proposed solutions for this problem exploit the diversity induced by antenna array or time oversampling, leading to the so-called, second order statistics techniques. Adaptive blind channel identification techniques based on a off-line least-squares approach have been proposed but this method assumes noise-free case. The method resorts to an adaptive filter with a linear constraint. This paper proposes a new approach based on eigenvalue decomposition. Indeed, the eigenvector corresponding to the minimum eigenvalue of the covariance matrix of the received signals contains the channel impulse response. And we present a adaptive algorithm to solve this problem. The performance of the proposed technique is evaluated over real measured channel and is compared to existing algorithms.

We propose a subchannel power control scheme in the OFDM system, which transmits data with a variable power level for each subchannel based on the received SNR. The OFDM system, employing the D-QPSK modulation and the proposed subchannel power control with a grouping coefficient equal to 3, gives about 2.3 dB gain in Eb/N0 comparing with the conventional OFDM system, under the two-ray multipath channel with the mean value of the second-ray's attenuation coefficient equal to 0.25, for the required BER equal to 10-5.

We have proposed the high-density triangular parallel wire channel MOSFET on an SOI substrate and demonstrated the suppressed short channel effects by simulation and experiment. In this device structure, the fabrication process is fully compatible with the planar MOSFET process and is much less complicated than other non-planer device structures including gate-all-around (GAA) and double-gate SOI MOSFETs. In addition, our fabrication process makes it possible to double the wire density resulting in the higher current drive. The three-dimensional simulation results show that the proposed triangular wire channel MOSFET has better short channel characteristics than single-gate and double-gate SOI MOSFETs. The fabricated triangular parallel wire channel MOSFETs show better subthreshold characteristics and less drain induced barrier lowering (DIBL) than the single-gate SOI MOSFETs.

This paper presents the analysis of integrated voice and data cellular networks with channel borrowing. Our considered system gives higher priority to handoff calls over new calls from users' point of view and reflects each characteristics of voice and data traffic types. Data handoff calls can wait in a queue while they are in handoff areas if there are no channels available. Voice handoff calls can borrow at most l channels from data calls if there are no idle channels upon their arrivals. We mathematically model this system by applying queueing theory. Then, we analyze its performance to derive the forced termination probability of data handoff calls, the blocking probabilities of the new and handoff calls of voice and data, and the Laplace Stieltjes transform for the distribution of waiting time in a queue. In numerical results, the analytical results for the mean waiting time of data handoff calls are compared with the simulation results to validate our analytical approach. Our system is also compared with the system where channel borrowing cannot be allowed (nonborrowing system) with respect to the blocking probabilities of the new and handoff calls of voice and data, the forced termination probability of data handoff calls, the mean and the coefficient of variation of the waiting time of data handoff calls.

A method for estimating propagation characteristics is described that uses the characteristics of pilot-data-inserted orthogonal frequency division multiplexing (OFDM) signal and is suitable for high-mobility OFDM transmission scheme. Several pilot data are inserted periodically along the frequency axis before the inverse fast Fourier transformation (IFFT) process in the transmitter. At the receiver, the received OFDM signal is correlated with a prepared distinctive OFDM signal in which several pilot data are inserted in the same positions as in the transmitted OFDM symbols and zeros are inserted in the other positions. The propagation characteristics can be estimated precisely and used to cancel any interference caused by delayed waves. Computer simulation shows that this method can estimate the propagation characteristics, which can then be used to cancel the interference caused by delayed waves before the FFT at the receiver under fast multipath fading conditions.

Since components faults occurring at arbitrary places (primarily on the links) affect seriously network performance and reliability, the multicomputers operating in harsh environments should be designed to guarantee normal network-missions in presence of those faults. One solution to the end is a fault-tolerant routing scheme, which enables messages to safely reach their destinations avoiding failed links when transmission of messages is blocked by certain faults. In the paper, we develop a fault-tolerant routing algorithm with deadlock freedom in an n-dimensional meshed network, and validate its efficiency and effectiveness through proper simulations. The aspects of fault-tolerance is adopted by appending partial-adaptiveness and detouring to the e-cube algorithm, while using a wormhole routing for the backbone routing method. The phenomenon of deadlock incurred due to its adaptiveness is eliminated by classifying a physical channel into a couple of virtual channels.

This paper discusses the application of a fuzzy-ARTMAP neural network to digital communications channel equalization. This approach provides new solutions for solving the problems, such as complexity and long training, which found when implementing the previously developed neural-basis equalizers. The proposed fuzzy-ARTMAP equalizer is fast and easy to train and includes capabilities not found in other neural network approaches; a small number of parameters, no requirements for the choice of initial weights, automatic increase of hidden units, no risk of getting trapped in local minima, and the capability of adding new data without retraining previously trained data. In simulation studies, binary signals were generated at random in a linear channel with Gaussian noise. The performance of the proposed equalizer is compared with other neural net basis equalizers, specifically MLP and RBF equalizers.