In this paper, we propose a scheme of frequency sub-band allocation to obtain maximum throughput in an orthogonal frequency division multiple access (OFDMA) system where each user has a finite number of packets to transmit, which are generated from packet calls with arbitrary size and arbitrary arrival rate. The proposed scheme is evaluated in terms of throughput and user fairness in comparison with the proportional fairness (PF) scheme and the Greedy scheme under the finite queue length condition. Numerical results show that the proposed scheme is superior to the Greedy scheme in terms of both throughput and fairness for finite queue length.

In this paper, we formulate a plan to operate multi-hop relays in IEEE 802.22-based cognitive radio (CR) systems and evaluate system performance to consider the propriety of a multi-hop relay scheme in CR systems. A centralized radio resource management and a simple deployment of relay stations (RSs) are assessed to make relay operations feasible under CR conditions. Simulation results show that the proposed multi-hop relay scheme significantly increases system throughput compared to a no-relay CR system as the incumbent user (IU) traffic gets heavier. Furthermore, the optimal number of hops can be determined given the traffic conditions.

We propose an iterative channel decoding scheme for two or more multiple correlated sources. The correlated sources are separately turbo encoded without knowledge of the correlation and transmitted over noisy channels. The proposed decoder exploits the correlation of the multiple sources in an iterative soft decision decoding manner for joint detection of each of the transmitted data. Simulation results show that achieved performance for the more than two sources is also close to the Shannon and Slepian-Wolf limit and large additional SNR gain is obtained in comparison with the case of two sources. We also verify through simulation that no significant penalty results from the estimation of the source correlation in the decoding process and the code with a low error floor achieves good performance for a large number of the correlated sources.

In this paper, we present a new dependency parsing method for languages which have very small annotated corpus and for which methods of segmentation and morphological analysis producing a unique (automatically disambiguated) result are very unreliable. Our method works on a morphosyntactic lattice factorizing all possible segmentation and part-of-speech tagging results. The quality of the input to syntactic analysis is hence much better than that of an unreliable unique sequence of lemmatized and tagged words. We propose an adaptation of Eisner's algorithm for finding the k-best dependency trees in a morphosyntactic lattice structure encoding multiple results of morphosyntactic analysis. Moreover, we present how to use Dependency Insertion Grammar in order to adjust the scores and filter out invalid trees, the use of language model to rescore the parse trees and the k-best extension of our parsing model. The highest parsing accuracy reported in this paper is 74.32% which represents a 6.31% improvement compared to the model taking the input from the unreliable morphosyntactic analysis tools.

This paper presents a field penetration characteristic, into a metallic wall with a narrow slot, due to a nearby dipole source. Coupled integral equations are derived and solved by applying Galerkin's method of moments (MoM) for calculating the penetrating electric field. It is shown that the 26 dB attenuation level field penetration into the narrow slot occurs at the dipole source position of much more than about 0.6 λ separation distance along the lateral direction from the slot center. It is also found that the 30 dB attenuation levels field penetration appear at the observation positions of much more than about 0.5 λ along the direction of the slot length and about 1.07 λ separation distance along the direction of the slot width from the slot center.

In this paper, we focus on resource allocation schemes for minimizing the energy consumption of subscriber stations (SSs) in uplink flows of the IEEE 802.16 OFDMA systems. The resource allocation schemes assign subcarriers, powers, and data rates to each SS based on the measured signal to noise ratio (SNR) of the uplink channel and predefined modulation and coding scheme as system parameters. Previous research efforts to optimize resource allocation focus on the rate and throughput maximizations, and develop suboptimal heuristic algorithms. However, this paper intends to reduce the energy consumption of SSs by considering the relationship between energy efficiency and resource allocation. In order to clearly formulate the relationship, we use the Multiple Choice Knapsack (MCK) problem, which is proved to be an NP-hard problem. We propose two heuristic schemes to solve the NP-hard problem, which adaptively use the modulation and coding scheme, defined in the IEEE 802.16 OFDMA systems to minimize the required transmission power of each SS. Our simulation results show that the proposed schemes can reduce the energy consumption by up to 53% compared to the channel state information (CSI) scheme, which determines the modulation and coding level only considering the channel state information.

In this paper, we propose a modified scheme of MSODB and PMS, called Predictive User Mobility Behavior (PUMB) to improve performance of resource reservation and call admission control for cellular networks. This algorithm is proposed in which bandwidth is allocated more efficiently to neighboring cells by key mobility parameters in order to provide QoS guarantees for transferring traffic. The probability is used to form a cluster of cells and the shadow cluster, where a mobile unit is likely to visit. When a mobile unit may change the direction and migrate to the cell that does not belong to its shadow cluster, we can support it by making efficient use of predicted nonconforming call. Concomitantly, to ensure continuity of on-going calls with better utilization of resources, bandwidth is borrowed from predicted nonconforming calls and existing adaptive calls without affecting the minimum QoS guarantees. The performance of the PUMB is demonstrated by simulation results in terms of new call blocking probability, handoff call dropping probability, bandwidth utilization, call successful probability, and overhead message transmission when arrival rate and speed of mobile units are varied. Our results show that PUMB provides the better performances comparing with those of MSODB and PMS under different traffic conditions.

A coordinate plane representation of the resource requirements of digital modulation methods is presented, and an overall resource efficiency measure is proposed. This measure can be used for the comparison of digital modulation methods and the evaluation of an emerging modulation technique. Several typical digital modulation methods are compared based on this measure to show its validity.

A channel code is constructed using sparse matrices for stationary memoryless channels that do not necessarily have a symmetric property like a binary symmetric channel. It is also shown that the constructed code has the following remarkable properties. 1. Joint source-channel coding: Combining channel code with lossy source code, which is also constructed by sparse matrices, a simpler joint source-channel code can be constructed than that constructed by the ordinary block code. 2. Universal coding: The constructed channel code has a universal property under a specified condition.

When handover events occur during the transmission of multimedia traffic, efficient handover control procedures and radio resource allocation are necessary to maintain the same QoS of transmitted multimedia traffic because the QoS may be degraded by additional delay and information loss. In this paper we propose a new handover control method for the next generation mobile multimedia networks, in which the handover setup process is done in advance of a handover request by predicting the handover cell from mobile terminal's current position. The handover procedures for real-time sessions are performed based on the handover cell information and the resource reservation condition. The radio resources in the estimated adjacent cells should be reserved and allocated to guarantee the continuity of the real-time sessions. We conduct a simulation model that is focused on the handover failure rate and packet loss rate. The simulation results show that our proposed method provides better performance than the previous methods.

A fully integrated CMOS wideband Low Noise Amplifier (LNA) operating over 2.3-7 GHz is designed and fabricated using a 0.18 µm CMOS process. The proposed structure is a common source-common source (CS-CS) cascode amplifier with a coupling capacitor. It realizes both low voltage drop at load resistor (Rload) and high gain over 2.3-7 GHz with simultaneous noise and input matching and low power consumption. This paper presents the proposed design technique of a wideband LNA, and verifies its performance by simulation and measurement. This wideband LNA achieves an average gain (S21) of 16.5 (dB), an input return loss (S11) less than -8 dB, a noise figure (NF) of 3.4-6.7 dB, and a third order input interception point (IIP3) of -7.5-3 dBm at 2.3-7 GHz with power consumption of 10.8 mW under 1.8 V VDD.

In this paper, we focus on the adaptive resource allocation issue for uplink OFDMA systems. The resources are allocated according to a proportional fairness criterion, which can strike an alterable balance between fairness and efficiency. Optimization theory is used to analyze the multi-constraint resource allocation problem and some heuristic characteristics about the optimal solution are obtained. To deal with the cohesiveness of the necessary conditions, we resort to bargaining theory that has been deeply investigated in game theory. Firstly, we summarize some assumptions about bargaining theory and show their similarities with the resource allocation process. Then we propose a priority-ranked bargaining model, whose primary contribution is applying the economic thought to the resource allocation process. A priority-ranked bargaining algorithm (PRBA) is subsequently proposed to permit the base station to auction the subcarriers one by one according to the users' current priority. By adjusting the predefined rate ratio flexibly, PRBA can achieve different degrees of fairness among the users' capacity. Simulation results show that PRBA can achieve similar performance of the max-min scheme and the NBS scheme in the case of appropriate predefined rate ratio.

Multi-source broadcasting is one of the information dissemination problems on communication networks such that some units disseminate distinct messages to all other units. In this paper, we study multi-source broadcasting on the de Bruijn and Kautz digraphs which are the models of interconnection networks. In [8] and [12], a cycle-rooted tree which has a large root-cycle is constructed by composition of isomorphic factors, and the multi-source broadcasting is executed on the cycle-rooted tree. On the other side, we execute multi-source broadcasting on each isomorphic factors at the same time. We present a method for multi-source broadcasting using isomorphic cycle-rooted trees which factorize these digraphs, and investigate its efficiency.

In this paper, we analyze the extended real-time Polling Service (ertPS) algorithm in IEEE 802.16e systems, which is designed to support Voice-over-Internet-Protocol (VoIP) services with data packets of various sizes and silence suppression. The analysis uses a two-dimensional Markov Chain, where the grant size and the voice packet state are considered, and an approximation formula for the total throughput in the ertPS algorithm is derived. Next, to improve the performance of the ertPS algorithm, we propose an enhanced uplink resource allocation algorithm, called the e 2rtPS algorithm, for VoIP services in IEEE 802.16e systems. The e 2rtPS algorithm considers the queue status information and tries to alleviate the queue congestion as soon as possible by using remaining network resources. Numerical results are provided to show the accuracy of the approximation analysis for the ertPS algorithm and to verify the effectiveness of the e 2rtPS algorithm.

When the joint source-channel (JSC) decoder is used for source coding over noisy channels, the JSC decoder may invent errors even though the received data is not corrupted by the channel noise, if the JSC decoder assumes the channel was noisy. A novel encoder algorithm has been recently proposed to improve the performance of the communications system under this situation. In this letter, we propose another algorithm based on conditional entropy-constrained vector quantizer to further improve the encoder. The algorithm proposed in this letter significantly improves the performance of the communications system when the hypothesized channel bit error rate is high.

The 3GPP LTE-Advanced has been attracting much attention recently, where the channel bandwidth would be beyond the maximum bandwidth of LTE, 20 MHz. In LTE, single carrier-frequency division multiple access (SC-FDMA) was accepted as the uplink access scheme due to its advantage of very low cubic metric (CM). For LTE-A wideband transmission, multicarrier access would be more effective than single carrier access to make use of multi-user diversity and can maintain the physical channel structure of LTE, where the control information is transmitted on the edges of each 20 MHz. In this paper, we discuss the access schemes in bandwidth under 20 MHz as well as over 20 MHz. In the case of bandwidth under 20 MHz, we propose the access schemes allowing discontinuous resource allocation to enhance average throughput while maintaining cell-edge user throughput, that is, DFT-spread-OFDM with spectrum division control (SDC) and adaptive selection of SC-FDMA and OFDM (SC+OFDM). The number of discontinuous spectrums is denoted as spectrum division (SD). For DFT-S-OFDM, we define a parameter max SD as the upper limit of SD. We evaluate our proposed schemes in bandwidth under 20 MHz and find that SC+OFDM as well as SDC with common max SD or UE-specific max SD can improve average throughput while their cell-edge user throughput can approach that of SC-FDMA. In the case of bandwidth over 20 MHz, we consider key factors to decide a feasible access scheme for aggregating several 20 MHz-wide bands.

Biologically-inspired approaches are one of the most promising approaches to realize highly-adaptive distributed systems. Biological systems inherently have self-* properties, such as self-stabilization, self-adaptation, self-configuration, self-optimization and self-healing. Thus, the application of biological systems into distributed systems has attracted a lot of attention recently. In this paper, we present one successful result of bio-inspired approach: we propose distributed algorithms for resource replication inspired by the single species population model. Resource replication is a crucial technique for improving system performance of distributed applications with shared resources. In systems using resource replication, generally, a larger number of replicas lead to shorter time to reach a replica of a requested resource but consume more storage of the hosts. Therefore, it is indispensable to adjust the number of replicas appropriately for the resource sharing application. This paper considers the problem for controlling the densities of replicas adaptively in dynamic networks and proposes two bio-inspired distributed algorithms for the problem. In the first algorithm, we try to control the replica density for a single resource. However, in a system where multiple resources coexist, the algorithm needs high network cost and the exact knowledge at each node about all resources in the network. In the second algorithm, the densities of all resources are controlled by the single algorithm without high network cost and the exact knowledge about all resources. This paper shows by simulations that these two algorithms realize self-adaptation of the replica density in dynamic networks.

This paper proposes block-wise resource block (RB)-level distributed OFDMA transmission with ND-block division in order to obtain the frequency diversity effect even for low-rate traffic (here ND indicates the number of virtual RBs within one physical RB) in Evolved UTRA downlink. More specifically, we propose a constraint rule such that distributed transmission is multiplexed into a different physical RB from that of localized transmission in order to achieve the same resource assignment and independent decoding between the distributed and localized transmissions. Based on the proposed rule, a virtual RB for distributed transmission is segmented into ND blocks with the size of 1/ND of the original virtual RB. Then, the ND virtual blocks with the size of 1/ND are mapped together into each ND physical RB in a distributed manner, resulting in a large frequency diversity effect. Numerical calculations show that the block-wise RB-level distributed transmission can reduce the number of control signaling bits required for resource assignment compared to the subcarrier-level distributed transmission scheme, which provides the best performance. Moreover, a system-level simulation shows that the loss in the cell throughput employing the block-wise RB-level distributed transmission compared to that using the subcarrier-level transmission is only within 3-4% when the channel load is 0.5 and 1.0, i.e., the maximum loss is 3-4% at approximately 90% in the cumulative distribution function (CDF).

An effective operation mode and a space-time synchronization technique for the spaceborne/airborne hybrid bistatic synthetic aperture radar (SA-BSAR) using sources of opportunity are presented. Our motivation lies in the fact that the existing approaches in the literature, where the transmitter antenna must be steered, can only be used in the hybrid bistatic SAR systems with cooperative transmitter. The presented mode is to widen the receiving beam for the purpose to increase the scene extension in azimuth. The inspiration comes from the much shorter receiving distance as compared to the one in mono-static spaceborne SAR. This means that the receiving gain can be significantly reduced to provide the same signal-to-noise ratio (SNR) with respect to the mono-static case. The feasibility of the wide-beam mode is first preliminarily verified by a quantitative analysis of SNR and a demonstration that the pulse repetition frequency (PRF) used in the spaceborne illuminator can easily satisfy the PRF constraints of the SA-BSAR. The influence on the azimuth ambiguity to signal ratio (AASR) is also discussed and the corresponding broadening factor of the maximum allowable for receiver beamwidth is subsequently derived. Afterwards, the formulae for calculating the overlap time, the scene extension and the azimuth resolution are deduced. As there are no grating lobes in satellite antenna pattern since the non-cooperative illuminator normally operates in the side-looking mode, an existing technique for the space-time synchronization in cooperative hybrid systems can not be directly applied. The modification performed and its underlying principle are presented in detail. The simulation results demonstrate the effectiveness of the wide-beam mode, and show that in most cases a useful scene extension (on the order of at least 1 km) can be achieved with a roughly equivalent azimuth resolution as compared to the one in mono-static spaceborne SAR. In some cases, explicit measures to suppress the azimuth ambiguity must be taken to achieve the expected scene extension.

A sound field reproduction method is proposed that uses blind source separation and a head-related transfer function. In the proposed system, multichannel acoustic signals captured at distant microphones are decomposed to a set of location/signal pairs of virtual sound sources based on frequency-domain independent component analysis. After estimating the locations and the signals of the virtual sources by convolving the controlled acoustic transfer functions with each signal, the spatial sound is constructed at the selected point. In experiments, a sound field made by six sound sources is captured using 48 distant microphones and decomposed into sets of virtual sound sources. Since subjective evaluation shows no significant difference between natural and reconstructed sound when six virtual sources and are used, the effectiveness of the decomposing algorithm as well as the virtual source representation are confirmed.