The effect of feedback delay and channel estimation error on closed-loop transmit diversity (CTD) systems is investigated in time-selective Rayleigh fading channels. Based on a minimum mean square error (MMSE) channel estimator, the variance of the estimation error is formulated in terms of fading index and the number of transmit antennas. A bit error rate (BER) expression for the CTD system is analytically derived as a function of channel estimation error, feedback delay, and fading index. It is shown that the BER performance of the CTD system improves as the length of training symbols increases and/or the frame length decreases. In the CTD system, more accurate channel estimation scheme is required to achieve its full gain as the number of employed transmit antennas increases. It is also found that the CTD system is applicable to the slowly moving channel environments, such as pedestrians, but not for fast moving vehicles.

This paper is concerned with the signal processing aspects of the recently proposed interleave-division multiple-access (IDMA) scheme. We propose several low-cost detection algorithms to solve the problems of multiple-access, cross-antenna and intersymbol interference (ISI). The complexities (per user) of these algorithms are very low and increase either linearly or quadratically with the number of paths. It is shown that an IDMA system with a rate- 16-state convolutional code and a length-8 spreading sequence can support more than 100 users in a multipath fading channel with two receive antennas. This clearly indicates the great potential of IDMA systems.

This letter presents an investigation of channel estimation scheme for a high rate WPAN system using multiple transmit antennas over indoor wireless channel. A simple algorithm utilizing the autocorrelation property of a CAZAC preamble is proposed for channel estimation. Simulation and analytical results show the performance of the proposed algorithm in terms of mean square error (MSE) of channel estimation. At the same time, the effect of imperfect channel estimation introduced by relatively large RMS delay spread is highlighted.

In this paper, we consider a blind channel estimation and equalization for single input multiple output (SIMO) channels. It is based on the one-step forward multichannel linear prediction error method. The derivation of the existing method is based on the noiseless assumption, however, we analyze the effects of additive noise at the output of the one-step forward multichannel linear prediction error filters. Moreover, we derive analytical results for the error in the blind channel estimation and equalization using linear prediction.

This paper investigates a random-interleaver-based approach to space-time coding. The basic principle is to employ a good forward error correction (FEC) code and transmit randomly interleaved codewords over an antenna array. A low-cost estimation technique is considered. The complexity involved grows only linearly with the number of transmit antennas. Near-capacity performance can be achieved with moderate complexity.

In this paper, we propose an algorithm, named efficient utilization polling (EUP), to support asynchronous data traffic at MAC layer by using the characteristics of Bluetooth technology. The algorithm uses a single bit in the payload header to carry the knowledge of queues in slaves for dynamically adapting the polling intervals for achieving the goals of high channel utilization and power conserving. In addition, we propose a differentiation mechanism, named shift-polling window (SPW). Based on EUP, the SPW differentiates the throughput from various classes, and still keeps the link utilization high and almost the same as that of the best-effort services. Extensive simulations are experimented on the behavior of the EUP and SPW by tuning the related parameters, such as polling interval, buffer size, queue threshold level, etc., in order to verify the expectation of these methods.

In packet networks including the Internet and commercial 3G wireless bearers, the network states that a streaming media application experiences are not known a priori and exhibit time-varying characteristics. For such dynamic environments, network-adaptive techniques are essential to efficiently deliver video data. In this paper, we propose a frame-based optimal scheduling algorithm which incorporates a MAP (Maximum A Posteriori) framework for adapting to varying network loss rate. The optimal transmission schedule is determined such that effective frame-rate is maximized at playback. Also, for multiple packets per frame, frame-based selection of delivery order greatly reduces computational complexity for a server scheduler when compared with packet-based scheduling techniques. In addition, by dynamically estimating instantaneous packet loss probability, the proposed scheduler performs network-adaptive transmission for streaming video over time-varying packet networks. Simulation results for test video sequence show that the proposed scheduling algorithm outperforms conventional ARQ-based schemes from a view point of reconstructed video quality as well as playable frame-rate. In particular, the proposed scheduling algorithm exhibits significant improvements of frame-rate over highly lossy channels.

This paper discusses the implementation of multi-band digital intermediate frequency (IF) for wideband CDMA (W-CDMA) transceiver. The majority of the implemented module in hardware is composed of wideband analog-to-digital converters (ADCs), digital-to-analog converters (DACs), and field programmable-gated-arrays (FPGA). And in software, it is coded by VHSIC hardware description language (VHDL) for realizing digital filters and numerically controlled oscillator, etc. To cope with the hardware limitation such as the number of gates in FPGA, the overall digital filter embedded in transceiver is constructed via a cascading a series of decimation and interpolation filters. At transmitter, in order to upconvert the multi-band baseband channels simultaneously, two-stage digital complex quadrature modulation (DCQM) is utilized. The relevant up-and-down conversion of the numerically controlled oscillator (NCO) is designed in the form of a look-up-table (LUT), having samples associated with a sampled sinusoidal with period of 1/4. At receiver, to avoid the usage of surface acoustic wave (SAW) filter, the high-performance digital filter is implemented subject to satisfying band rejection ratio prescribed in blocker and adjacent channel specification. This paper provides the performance of the implemented digital IF module by revealing the results taken from the measurement instruments. Moreover, to confirm its validity computer simulations are simultaneously conducted.

In providing video programs to a number of clients through networks, a broadcasting approach is more appropriate than a true-video-on-demand approach in efficient use of bandwidth. However, the broadcasting approach also needs excessively wide bandwidth if many video programs are to be transmitted. This study presents a very simple but novel architecture called dynamic channel broadcasting, for video-on-demand systems. The proposed architecture uses both static and dynamic broadcasting channels to improve the efficiency of channels. The proposed architecture eliminates the necessity of dividing each video into segments and switching channels frequently unlike the pyramid broadcasting, the skyscraper broadcasting and the harmonic broadcasting. Also this new architecture needs a smaller buffer size. The numerical results demonstrate that the newly proposed approach in some cases requires only 14% of the bandwidth required for the conventional broadcasting while maintaining the start-up latency.

In this study, a fourth-order cumulants based iterative algorithm for blind channel equalization is introduced, which is robust with respect to the existence of heavy Gaussian noise in a channel and does not require the minimum phase characteristic of the channel. The transmitted signals at the receiver are over-sampled to ensure the channel described by a full-column rank matrix. It changes a single-input/single-output (SISO) finite-impulse response (FIR) channel to a single-input/multi-output (SIMO) channel. Based on the properties of the fourth-order cumulants of the over-sampled channel inputs, the iterative algorithm is derived to estimate the deconvolution matrix which makes the overall transfer matrix transparent, i.e., it can be reduced to the identity matrix by simple reordering and scaling. In simulation studies, both a closed-form and a stochastic version of the proposed algorithm are tested with three-ray multi-path channels, and their performances are compared with the methods based on conventional second-order statistics and higher-order statistics (HOS) as well. Relatively good results with fast convergence speed are achieved, even when the transmitted symbols are significantly corrupted with Gaussian noise.

This paper presents a novel decoding strategy called combined iterative demapping/decoding (CIDD), for coded M-ary biorthogonal keying-based direct sequence ultra-wideband (MBOK DS-UWB) systems. A coded MBOK DS-UWB system consists of a convolutional encoder, an interleaver, and an MBOK DS-UWB pulse mapper. CIDD improves the error rate performance of MBOK DS-UWB systems by applying the turbo principle to the demapping and decoding processes at the receiver side. To develop the CIDD, a soft-in/soft-out MBOK demapping algorithm, based on the max-log-MAP algorithm, was derived. Simulation results showed that using CIDD siginificantly improved the error rate performance of both static and multipath fading channels. It was also shown that the computational complexity of CIDD is comparable to that of the Viterbi decoding used in [133,171]8 conventional convolutional coding.

One of the categories of decoding techniques for DFT codes in erasure channels is the class of iterative algorithms. Iterative algorithms can be considered as kind of alternating mapping methods using the given information in a repetitive way. In this paper, we propose a new iterative method for decoding DFT codes. It will be shown that the proposed method outperforms the well-known methods such as Wiley/Marvasti, and ADPW methods in the decoding of DFT codes in erasure channels.

We propose use of QR factorization with sort and Dijkstra's algorithm for decreasing the computational complexity of the sphere decoder that is used for ML detection of signals on the multi-antenna fading channel. QR factorization with sort decreases the complexity of searching part of the decoder with small increase in the complexity required for preprocessing part of the decoder. Dijkstra's algorithm decreases the complexity of searching part of the decoder with increase in the storage complexity. The computer simulation demonstrates that the complexity of the decoder is reduced by the proposed methods significantly.

In mobile cellular networks it is crucial to be able to use the available radio spectrum as efficiently as possible while providing a certain level of Quality of Service (QoS) for the users. Emergence of miscellaneous services has dramatically increased the complexity of this problem by creating a heterogeneous traffic environment. In this paper an efficient resource allocation scheme between two real-time services with different bandwidth requirements has been proposed for cellular networks with multimedia offered traffic and highly mobile users, which combines classical intera-cell resource borrowing concept with a novel inter-cell resource-sharing scheme between diverse classes of traffic. By assuming the heterogeneous offered traffic to be a combination of audio and video traffic types, through extensive simulations it will be shown that HCBA-UCB is capable of significantly improving audio teletraffic performance of the system while preserving fairness in service provision, i.e. without imposing additional expenses upon video QoS performance.

We design a DAB system using space-time coding for reliable transmission in the multipath fading channel. This letter assess the performance evaluation of space-time coded DAB system. For channel estimation in the space-time coded DAB system, we introduce two training sequences and derive the mean square error of each training sequence. We then represent the average symbol error rate of the space-time coded DAB system for the specific cases of two, three, four transmit antennas.

Sensor networks, in which a large number of sensor nodes are connected with each other through communication networks, are potential to provide extensive new applications with various research and technical challenges and attracting much attention all over the world. In this paper we describe information and signal processing for sensor networks with emphasis on the concepts of collaboration and fusion as one of the most important issues for the sensor networks. We also review some basic aspects of sensor networks in terms of their features and the network architecture.

Orthogonal Frequency Division Multiplexing (OFDM) is a promising technique for achieving high bit-rate transmission in radio environments. Various techniques to estimate channel attenuation have been proposed for OFDM transmission. In these techniques, the added pilot semi-blind (APSB) channel estimation has been proposed, which does not require any additional bandwidth. For an OFDM packet transmission that includes preambles, it is difficult to track the channel variation of the latter of packet due to time-varying channels in fast fading environments. We expect that the channel estimation is improved by applying the APSB channel estimation technique to the last symbols of packets without any additional bandwidth or degradation of bit rate. In this paper, we propose an OFDM packet transmission system with APSB channel estimation technique where this technique is applied to the last symbols of packets. We expect that the proposed system can track the fast time-variant channels without decrease of the data-rate, and the packet error rate (PER) is improved. We show that our proposed system is effective for improving the accuracy of the channel estimate in fast fading channels.

In December 2001, the authors started two kinds of experiments on the meteor burst communication (MBC) in Antarctica to study the ability of MBC as a communication medium for data collection systems in that region. In the first experiment, a continuous tone signal is transmitted from Zhongshan Station. The signal received at Syowa Station (about 1,400 km apart) is recorded and analyzed. This experiment is aimed to study basic properties of the meteor burst channel in that high latitude region. On the other hand, the second experiment is designed to estimate data throughput of a commercial MBC system in that region. A remote station at Zhongshan Station tries to transfer data packets each consisting of 10 data words to the master station at Syowa Station. Data packets are generated with five minutes interval. In this paper, we explain the experiments, briefly examine the results of the first year (from April 2002 to March 2003), and put forward the plan for the experiments in the second and third year. From the data available thus far, we can see that 1) the sinusoidal daily variation in the meteor activity typical in middle and low latitude regions can not be clearly seen, 2) non-meteoric propagations frequently dominate the channel especially during night hours, 3) about 60% of the generated data packets are successfully transferred to the master station within two hours delay even though we are now operating the data transfer system only for five minutes in each ten minutes interval, etc.

For future high-speed wireless communications using orthogonal frequency division multiplexing (OFDM), two major system requirements will emerge: throughput improvement and rich interference elimination. Because of its broadband nature and limited frequency allocations worldwide, interference from co-located wireless LAN's operating in the same frequency band will become a serious deployment issue. Adaptive array antenna can enhance the performance by suppressing the co-channel interference even when interference may have a large amount of multipath and also have similar received power to the desired signal. There are typically two types of adaptive array architecture for OFDM systems, whose signal processing is carried out before or after FFT (Fast Fourier Transform). In general, the pre-FFT array processing has low complexity, but in rich multipath and interference environments, the performance will deteriorate drastically. In contrast, the post-FFT array processing can provide the optimum performance even in such severe environments at the cost of complexity. Therefore, complexity-reduction techniques combined with the achievement of high system performance will be a key issue for adaptive array antenna applications. This paper proposes novel adaptive array architecture, which is a complexity-reduction technique using subcarrier clustering for post-FFT adaptive array. In the proposed scheme, plural subcarriers can be clustered into a group with the same spatial weight. Simulation results show that the proposed architecture is a promising candidate for real implementation, since it can achieve high performance with much lower complexity even in a rich multipath environment with low signal to noise plus interference ratio (SNIR).

Unlike most of the previous work for smart antennas that covered each area individually (antenna-array design, signal processing and communications algorithms and network throughput), this paper may be considered as a review of comprehensive effort on smart antennas that examines and integrates antenna array design, the development of signal processing algorithms (for angle of arrival estimation and adaptive beamforming), strategies for combating fading, and the impact on the network throughput. In particular, this study considers problems dealing with the impact of the antenna design on the network throughput. In addition, fading channels and tradeoffs between diversity combining and adaptive beamforming are examined as well as channel coding to improve the system performance.