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.

The maximum a posteriori (MAP) algorithm is the optimum solution for decoding concatenated codes, such as turbo codes. Since the MAP algorithm is computationally complex, more efficient algorithms, such as the Max-Log-MAP algorithm and the soft-output Viterbi algorithm (SOVA), can be used as suboptimum solutions. Especially, the Max-Log-MAP algorithm is widely used, due to its near-optimum performance and lower complexity compared with the MAP algorithm. In this paper, we propose an efficient algorithm for decoding concatenated codes by modifying the Max-Log-MAP algorithm. The efficient implementation of the backward recursion and the log-likelihood ratio (LLR) update in the proposed algorithm improves its computational efficiency. Memory is utilized more efficiently if the sliding window algorithm is adopted. Computer simulations and analysis show that the proposed algorithm requires a considerably lower number of computations compared with the Max-Log-MAP algorithm, while providing the same overall performance.

Application of multipled block codes (MBCs) for realizing new flexible and efficient transmission systems that feature in hunting-free reframing and asynchronous multiplexing is investigated. First, the principles of MBC are overviewed to show the capacities of filterless clock recovery. Then it is shown that modification of simple frame structure of MBC line code can be used for attaining hunting-free reframing for multiplexing systems. Two types of MBCs are developed to this end. While the one uses header blocks for hunting-free reframing, the other uses distributed frame patterns. Header design of multipled block codes (MBC) for hunting-free reframing (HFR) is investigated for frame patterns with and without violation compensation. The feasibility of hunting-free reframing is tested in an experimental system. Application of hunting-free reframing to asynchronous multiplexing is also investigated and tested in an experimental system. Finally, advantages of hunting-free multiplexing systems are discussed.

Spectral-amplitude coding (SAC) techniques in fiber-Bragg-grating (FBG)-based optical code-division multiple-access (OCDMA) systems were investigated in our previous work. This paper adopts the same network architecture to investigate the simultaneous reductions of multiple-access interference (MAI) and optical beat interference (OBI). The MAI is caused by overlapping wavelengths from undesired network coder/decoders (codecs) while the OBI is induced by interaction of simultaneous chips at adjacent gratings. It is proposed that MAI and OBI reductions may be obtained by use of: 1) a source spectrum that is divided into equal chip spacing; 2) coded FBGs characterized by approximately the same number of "0" and "1" code elements; and 3) spectrally balanced photo-detectors. With quasi-orthogonal Walsh-Hadamard coded FBGs, complementary spectral chips is employed as signal pairs to be recombined and detected in balanced photo-detectors, thus achieving simultaneous suppression of both MAIs and OBIs. Simulation results showed that the proposed OCDMA spectral-amplitude coding scheme achieves significant MAI and OBI reductions.

Fairness is one of the most important features of a rate allocation strategy. Proportional fairness criterion has been recently proposed by F. P. Kelly and his colleagues. In this paper, we have proposed a two-level hierarchical technique which allocates proportionally-fair rates to the network elastic users. Part of the network links which are used commonly by the end-users and are congestion prone, constitute the higher (first) level of the hierarchy. In this level, the users with common path in the network are grouped as virtual users. End-users and remaining network links constitute the lower (second) level of hierarchy. To improve the convergence rate of the algorithm, a combination of Jacobi method and fuzzy techniques is deployed in the higher level of hierarchy. Implementing such fast algorithms in the higher level (which is topologically simpler than the whole network), reduces the computational complexity with respect to the use of such algorithms in the whole network. Additionally, the lower level penalty function computation is done once in each N iterations, which reduces the computational complexity furthermore. The simulation results show that the proposed algorithm outperforms that of Kelly in the convergence speed.

In this paper, we consider the problem of multicast routing and wavelength assignment (MC-RWA) in multi-fiber all-optical WDM networks. Two main network design system comprehensively investigated here are mesh and multi-ring designs. Given the multicast traffic demands, we present new ILP formulations to solve the MC-RWA problem with an objective to determine the minimal number of fibers needed to support the multicast requests. Unlike previous studies, our ILP formulations are not only capable of finding the optimal multicast routing and wavelength assignment pattern to the light-trees, but also finding the optimal light-tree structures simultaneously. Since broadcast and unicast communications are special cases of multicast communications, our ILP models are actually the generalized RWA mathematical models of optical WDM networks. In addition to proposing the ILP models, this paper takes two main issues affecting the network capacity requirement into account, that is, the splitting degree level of optical splitters and techniques of wavelength assignment to the light-trees. Three multicast wavelength assignment techniques studied in this paper are Light-Tree (LT), Virtual Light-Tree (VLT) and Partial Virtual Light-Tree (PVLT) techniques. Due to the NP-completeness of the MC-RWA problem, the ILP formulations can reasonably cope with small and moderate networks. To work with large networks, this paper presents alternative MC-RWA ILP-based heuristic algorithms for the PVLT and LT networks and develops lower bound techniques to characterize the performance of our algorithms. Using existing large backbone networks, numerical results are reported to analyze such aspects as multiple fiber systems, the benefits of using optical splitters and wavelength converters, and the capacity difference between the mesh and multi-ring designs. Finally, this paper provides an analysis of the influence of network connectivity on the network implementation under the constraints of mesh and multi-ring design schemes.

Active Reliable Multicast (ARM) is a novel loss recovery scheme for large-scale reliable multicast that employs active routers to protect the sender and network bandwidth from unnecessary feedback and repair traffic. Active routers perform NACKs suppression, cache multicast data for local loss recovery, and use scoped retransmission to avoid exposure. Limited active resources at routers need to be optimized to achieve low loss recovery latency and/or high network throughput. In this paper, we study the cache placement strategies and caching policies for ARM. Several heuristics, namely uniform allocation, proportional allocation, max-min fair share and weighted allocation for cache allocation methods are proposed. To further improve the loss recovery performance, caching policies can be employed in conjunction with the cache allocation strategies. Several caching policies, namely complete caching, random caching and deterministic caching, are proposed. Extensive simulation experiments are conducted to evaluate and compare the performance of the proposed strategies and policies. Numerical results reveal that significant performance gains can be achieved when a proper cache placement strategy and a caching policy are used for a given available cache resource. Another interesting finding is that the contributions of the cache placement scheme and caching policy to the recovery latency performance are roughly independent. The obtained insights in this study will provide some design guidelines for optimal active resource allocation and caching polices for reliable multicast communications.

In this paper, we term multimedia streaming application traffic "stream flows" and the other usual application traffic "non-stream flows." Many problems occur when both flows are aggregated on a shared link because the different TCP and UDP behaviors cause negative interactions. One way to solve these problems is to isolate stream and non-stream flows to different classes. However, it is difficult to determine the bandwidth allocation for each class and dynamic bandwidth allocation schemes are hard to implement on large scale networks. We therefore propose a dynamic class assignment method that maintains the QoS and that has a higher scalability than dynamic bandwidth allocation schemes. It is workable on Diffserv AF PHB. The outline is as follows. We classify non-stream flows into four classes and dynamically assign stream flows to the classes, taking the conditions and characteristics of the classes into consideration. On assigning classes to stream flows, we map them to a higher drop precedence than non-stream flows not to degrade the QoS of them, based on the assumption that occasional packets being dropped do not create serious problems for them. In this paper, we first discuss our classification of non-stream flows, and present the characteristics of non-stream flows in each class. We then discuss our drop precedence mapping. After this, we propose an algorithm for our method of dynamic class assignment and provide some simulation results where it could provide constant qualities with stream and non-stream flows, adapting to changing traffic.

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.

Next generation wireless networks are expected to support multimedia applications (audio phone, video on demand, video conference, file transfer, etc.). Multimedia applications make a great demand for bandwidth and impose stringent quality of service (QoS) requirements on the wireless networks. In order to provide mobile hosts with high QoS, efficient and better bandwidth reservation is necessary in multimedia wireless networks. This paper presents a novel hybrid dynamic-grouping bandwidth reservation scheme to support QoS guarantees in the next generation wireless networks. The proposed scheme is based on probabilistic resource estimation to provide QoS guarantees for multimedia traffic in cellular networks. We establish several reservation time-sections, called groups, according to the mobility information of mobile hosts (MHs) of each base station (BS). The amount of reserved bandwidth for each BS is dynamically adjusted for each reservation group. We use the hybrid dynamic-grouping bandwidth reservation scheme to decrease the connection-dropping probability (CDP) and connection-blocking probability (CBP), while increasing the bandwidth utilization. The simulation results show that the hybrid dynamic-grouping bandwidth reservation scheme provides less CDP and less CBP, and achieves high bandwidth utilization.

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.

The combination of OFDM and multiple antennas in either the transmitter or receiver is attractive to increase a diversity gain. However, multiple antennas system requires an antenna separation of 5-10 λ to keep the correlation coefficient below 0.7 for the space diversity, so this may be difficult to implement in a mobile station with high mobility. Recently, the polarization transmit diversity is considered in a mobile station. However, polarization transmit diversity requires twice transmit powers to compare with the conventional transmit diversity, since only vertically polar antenna cannot receive the horizontal signal components. In this paper, we express the cross correlation of each polarization antenna and the cross polarization discrimination (XPD) of multiple polarization antennas with simple model, and we propose an wideband OFDM using Alamouti coded heterogeneous polarization antennas for reducing the previous problem. From the simulated results, the proposed system shows better BER performance than that of the conventional STBC/OFDM.

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.

In this paper, a sub-optimal Rake receiver combined with a Wiener Filter is investigated for use in an indoor environment. Inner-Chip-interference is dominant when the application is indoors, so the inner-chip-interference rejection function becomes critical for the receiver. Pilot symbols in each slot are used for channel estimation and weight calculation of Rake combining through Wiener Filter. Compared to conventional combining which uses maximum ratio combining, Wiener combining using IRC (Interference rejection combining) achieves better ICI (Inner-chip-Interference) rejection. This paper clarified that the sub optimal Rake receiver using Wiener Filter is 4 dB better than the conventional Rake receiver under the indoor application.

The use of coplanar waveguide (CPW)-fed ultra-wideband antennas in applications of multi-system integration has been demonstrated in this paper. Spiral slot antennas and feeding structures were fabricated on the same plane of the substrate so that the circuit process and the position alignment could be simplified. A CPW-fed spiral slot antenna possessing the ultra-wideband characteristic is also suitable for integration with a monolithic microwave integrated circuit (MMIC) module. Variations of the measured initial resonant frequency due to substrate thickness, number of turns and slot width are discussed in this paper. In addition, two topologies of the CPW-fed spiral slot antenna were devised and measured to demonstrate the capability of lowering the initial resonant frequency by adding a circularly microstrip stub at the end of the feedline and placing a short pin to terminate the spiral slot and feedline. According to the measured results, the CPW-fed spiral slot antenna covered most of the commercial wireless communication and satellite communication systems in radio frequency (RF), microwave and millimeter-wave applications.

Imaging techniques for robots are important and meaningful in the near future. Pulse radar systems have a great potential for shape estimation and locationing of targets. They have an advantage that they can be used even in critical situations where optical techniques cannot be used. It is thus required to develop high-resolution imaging algorithms for pulse radar systems. High-resolution imaging algorithms utilize the carrier phase of received signals. However, their estimation accuracy suffers degradation due to phase rotation of the received signal because the phase depends on the shape of the target. In this paper, we propose a phase compensation algorithm for high-resolution pulse radar systems. The proposed algorithm works well with SEABED algorithm, which is a non-parametric algorithm of estimating target shapes based on a reversible transform. The theory is presented first and numerical simulation results follow. We show the estimation accuracy is remarkably improved without sacrificing the resolution using the proposed algorithm.

We have developed a quick battery checker the Li-ion battery packs used in mobile phones. It checks for capacity deterioration by using an impedance-measuring method. Our previous measurements of the capacity and impedance at 1 kHz for various battery packs proved conclusively that there is a strong correlation between degraded capacity and impedance. The battery checker's design took into account the results we obtained from measuring impedances. We showed that the battery checker is highly accurate and fast.

The MPEG-2 Test Model 5 (TM5) algorithm describes a rate control method which consists of three steps: bit allocation, rate control and modulation. In TM5, however, buffer overflow and picture quality degradation may occur at the end of the GOP because the target bits and the actual coding bits for each picture do not match well. This paper presents a new bit rate control algorithm for matching the target and the actual coding bits based on accurate bit allocation. The key idea of the proposed algorithm is to determine quantization parameters which enable us to generate the actual coding bits close to the target bits for each picture, while improving the picture quality. The proposed algorithm exploits the relationship between the number of the actual coding bits and the number of the estimated bits of the previous macroblock within a picture.

In this paper, we propose an optimal bit rate control algorithm which is fully compatible with MPEG-4 or H.263+. The proposed algorithm is designed to identify the optimal quantizer set through Lagrangian optimization when used for optimal bit allocation. To find the optimal quantizer set, we make use of the Viterbi algorithm in order to solve the dependency between quantization parameters of each macroblock due to the unique characteristics of MPEG-4 or H.263+. We set the Lagrangian cost function as a cost function of the Viterbi algorithm. We implement the proposed algorithm in MPEG-4 coders and compare its performance to the VM8 and optimal bit rate control algorithm, using independent quantization parameters in the circumstance of a low bit rate.

Voice over Internet protocol (VoIP) is to transfer voice packets over IP networks, while voice signal is processed by using digital signal processing technology before being transmitted. VoIP quality cannot be expected, because it is hard to predict the influence of delay, packet loss rate, packet error, etc. It is difficult to rebuild the voice wave form, if a large amount of voice packets are lost. This paper mainly studies on how to maintain a better voice quality over hybrid fiber/coaxial (HFC) networks, if it is inevitable to drop packets. We particularly consider the data over cable service interface specification (DOCSIS) version 1.1 with the unsolicited grant service with activity detection (UGS/AD) for VoIP services. We propose a smallest successive times first (SSTF) scheduling algorithm to schedule VoIP packets for cable modem termination system (CMTS), which can support fair transmission and long-term transmission continuity for VoIP connections. We analyze voice quality about continuity of the transmitted VoIP packets, consecutive clipping times, and VoIP packet drop rate for all connections. Performance measurement shows excellent results for the proposed algorithm by simulation experiments and objective evaluation.

Direct digital frequency synthesizers (DDFS) provide fast frequency switching with high spectral purity and are widely used in modern spread spectrum wireless communication systems. ROM-based DDFS uses a ROM lookup table to store the amplitude of a sine wave. A large ROM table is required for high spectral purity. However, a larger ROM uses more area and consumes more power. Several ROM compression methods, including Sunderland technique based on simple trigonometric identities and quantization & error compensation techniques, have been reported. In this paper, we suggest several new techniques to reduce the ROM size. One new technique uses more number of hierarchical levels in ROM structures. Another technique uses simple interpolation techniques combined with hierarchical ROM structures. Experimental results show that the new proposed techniques can reduce the required ROM size up to 24%, when compared to that of a resent approach.

In this letter, we propose to construct reliable overlay data delivery tree based on group member's packet loss rate while preserving end-to-end delay below predetermined threshold. Through practical simulation, performance is evaluated and compared.

To prevent performance degradation of TCP due to packet losses in the smooth handoff by the route optimization extension of Mobile IP standard, a few packet buffering methods have been proposed. The packet buffering at the BS recovers the packets dropped during an inter-subnetwork handoff, by forwarding the buffered packets at the previous BS to the new BS to which the mobile host is connected after handoff. However, when the mobile host user moves to a congested BS in a new foreign subnetwork, those buffered packets are likely to be dropped at the new BS. Thus, as well as the TCP connections of the mobile host which have moved into the new BS, the already existing TCP connections of the new BS experience severe performance degradation. This effect is due to the increased congestion by the forwarded burst packets; all of the TCP connections can initiate their congestion control algorithms simultaneously, i.e., global synchronization. This paper will consider a general case where a mobile host user moves into a congested BS of a new foreign subnetwork. We analyze the influence of the packet buffering on the TCP performance in the new BS, for the Drop-Tail and Random Early Detection (RED) buffers. Simulation results show that although the RED buffer gives better handoff performance than the Drop-Tail buffer, it cannot avoid a large decrease in the TCP throughputs due to global synchronization, when a TCP connection is added at the BS by an inter-subnetwork handoff. Finally, we discuss some methods that can address the negative effect of the packet buffering method.

With iterative turbo decoding, the reliability of each bit in a frame is not same after some iterations. We propose novel bit-level hybrid automatic repeat request (HARQ) schemes with turbo codes in which only the unreliable bit and its neighboring bits are retransmitted based on the decoding reliability. The proposed bit-level HARQ schemes improve error performance compared with conventional HARQ schemes.

Bluetooth is a system for providing short-range, small size, low-power and low-cost connectivity operating in the ISM (Industrial Scientific Medicine) band at 2.4 GHz. Bluetooth has been seen as a promising candidate for ad-hoc wireless networking and wireless personal area network (WPAN). In this paper, we first discuss previously proposed polling algorithms in Bluetooth piconet. We then propose an efficient fair scheduling algorithm which improves the throughput efficiency of the system by adaptively assigning the polling interval according to the number of inactive slaves. We also show the simulation results of the proposed algorithm compared with previously proposed algorithms.

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 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.

We propose a multiple-subcarrier (MS) optical communication system using intensity modulation with direct detection (IM/DD) with peak reduction carriers (PRCs) to improve the power efficiency of IM/DD MS systems. The proposed system transmits L subcarriers referred to as PRCs among N subcarriers for the d.c. bias reduction so that the optical power is reduced. Since information bits are mapped onto each subcarrier other than PRCs independently, the information bits of each subcarrier can be detected independently and the error rate of the proposed system is unaffected by PRCs.

Some conventional beamformers require the direction of the desired signal. The performance of such beamformers can substantially be degraded even in the presence of small error on the directional information. In this letter, we propose a prefilter-type beamforming scheme robust to directional error by employing a simple compensator. The performance of the proposed scheme is verified by computer simulation.

In this letter, a circular microstrip patch antenna with a conical cup is proposed. The results of a simulation and experiment show that the conical cup has a beneficial effect on the antenna's gain and principal plane beamwidths. The maximum gain of this antenna was 12.6 dBi, which is about 3 dB higher than one with a cylindrical cup. The 3-dB beamwidths of the E-and H-planes were 34and 44, respectively.

This letter develops a practical sectorized antenna array using center-fed half-wavelength dipole antennas that are parallel to and a distance in front of a large ground plane reflector. Each element in the array is designed to provide coverage to isolate each 120sector from adjacent sectors. We derive a closed-form expression for spatial correlation function that can be used as guides in evaluating the effects of array spatial correlation on diversity performance in sectorized cellular communications.

A compact dual feed integrated triple-band vehicle antenna for GPS and RKES/PCS operation is proposed. The GPS (1575.42 1.023 MHz) antenna element is a corner truncated microstrip patch providing circularly polarized broadside radiation pattern. The RKES/PCS (447.7375 MHz/1750-1870 MHz) antenna consists of two printed monopoles fed by coplanar waveguide (CPW) transmission line. To reduce the height of RKES antenna, modified inverted-L monopole structure is utilized. The proposed antenna satisfies all the bandwidth and gain requirements for GPS and RKES/PCS services. Good radiation characteristics for vehicle application have also been obtained.

A high accuracy numerical technique based on the Finite Difference Time Domain (FDTD) method for a long dipole antenna analysis is presented. An improvement of the accuracy can be achieved without reducing the cell size by incorporating a quasi-static field behavior into the FDTD update equations. A closed form of the quasi-static field is obtained from a low frequency limit of a sinusoidal current distribution. The validity of the proposed algorithm is confirmed even when the length of dipole antenna is longer than half wavelength by comparing the results with the Method of Moment.

This letter presents a novel video and audio PTSs self-adaptive interlace strategy in MPEG-2 transport stream. By adaptive regulating the relative position of audio and video access units in bit-stream according to their PTSs, the proposed strategy provides reliable video and audio synchronization.