The Ocean Internet of Things (OIoT) has promoted the development of the ocean devices, which generate a large amount of underwater acoustic data. The transmission of underwater acoustic data requires a large amount of spectrum resources. Aiming at the problems of improving the utilization rate and security of spectrum resources, an underwater acoustic spectrum sensing algorithm based on personalized federated learning (PFL) and RepViG is proposed. A security protection framework is established for underwater acoustic data based on PFL. The common features of each underwater acoustic data are extracted through the meta-model. Each client only needs to fine-tune the parameters of the meta-model based on the local underwater acoustic data to achieve a personalized model. Based on the improved RepViT and the improved ViG, a dual-branch sensing model of RepViG is designed. In the improved RepViT, we employ the Haar wavelet downsampling (HWD) module to retain the low-frequency and high-frequency detail features of the underwater acoustic signal through multi-resolution. And we employ Interactive Convolution Block (ICB) to capture the relationship between local features and global features through multi-scale dynamic convolution kernels. In the improved ViG, we adopt the lightweight Star-Blcok module to reduce feature redundancy and enhance sensing efficiency. Compared with other algorithms, the simulation results show that the detection probability is increased by 7.6%, and the false alarm probability is reduced by 7.5%.

We present a matrix factorization-based recommender system enhanced by parameter tuning and Gaussian noise modeling to improve robustness and generalization. Using the Amazon Product Reviews dataset, we achieve a validation MSE of 0.464 with k=2 latent features. Noise injection prevents overfitting, and implicit feedback improves accuracy. The results offer practical insights for scalable and stable recommendation systems.

Wireless sensing technology, analyzing signal propagation to sense environments, has advanced in smart homes, health monitoring and security. However, massive data and dynamic communication environments expose limitations in traditional methods: weak security, poor feature extraction, and high environmental dependency. To address these challenges in complex cross-domain scenarios, this paper proposes a collaborative secure framework FL-BLC, which integrates federated learning (FL) and blockchain (BLC), and designs the DB-SE-Yolov8 cross-domain intelligent sensing algorithm. The FL-BLC framework ensures tamper-proof transmission and data privacy by encrypting and hash-verifying locally trained model parameters before batch-writing them to the blockchain. The DB-SE-Yolov8 algorithm employs a dual-branch(DB) design: the upper branch employs average adaptive pooling for global features, while the lower branch employs the Squeeze-and-Excitation (SE) attention mechanism for attention features. A gating mechanism dynamically fuses multi-scale features, reducing complexity and enhancing accuracy across diverse scenarios. Compared with Dual-Attention CSI Network, Environment Independent and Joint Adversarial Domain Adaptation algorithms, DB-SE-Yolov8 significantly improves in-domain and cross-domain sensing performance. For cross-location and cross-orientation scenarios, the sensing accuracy is improved by 7.16% and 8.84%, while sensing efficiency is improved by 10.85% and 11.67%, respectively.

This article presents a robust 3-D Received Signal Strength Difference (RSSD) localization algorithm under mixed Gaussian noise in Underwater Wireless Sensor Networks (UWSNs) with Non-Line-Of-Sight (NLOS) paths. To mitigate the adverse effects, concurrent to absorption and path losses on accurate underwater localization, an Efficient RSSD-based Iterative Estimator (ERIE) in mixed Gaussian noise and NLOS environments is proposed. First, the corresponding non-convex problem in such environments is formulated, and the direct solution to this problem is not tractable unfortunately. Considering underwater acoustic signal attenuation, an RSSD-based min-max strategy is designed to transform it into a problem minimizing the worst-case loss, combined with the Huber cost function, constitutes a Huber function-based equivalent problem (H-ADMM) solved by Alternating Direction Method of Multipliers (ADMM). A compensation matrix is designed based on the H-ADMM solution to compensate for the bias introduced by the transformation, and the corresponding Cramér-Rao Lower Bound (CRLB) is derived to provide a performance benchmark. Numerical results indicate that the proposed approach achieves a higher localization accuracy than state-of-the-art methods.

The main advantage of usingasymmetricfix-free codes is the possibility ofbidirectional decodingwhich can improve decoding speed and error resilience. However, the cost of these benefits is higher redundancy and the need to design two different decoders.Symmetricfix-free codes have the same forward and backward decoders but the redundancy of an optimal code can be unlimited. Withweakly symmetricfix-free (WSF) codes, the forward and backward decoders are the same and the redundancy of the optimal WSF code for an arbitrary source is limited to 1.0817 bits in the worst case. In this paper, we show that this redundancy is much less on average, i.e., the average redundancy of an optimal WSF code is at most 0.1241 bits for alphabet sizes larger than 32. TheA*-based search algorithm is modified to obtain the optimal WSF code for the average distribution of monotone sources withnsymbols forn≤ 64. Based on these optimal codes and a new code product operator, some suboptimal WSF codes with many codewords are obtained. Then upper bounds on the average redundancy of optimal WSF codes are derived for arbitrarily largen. These bounds are almost the same as those derived recently for asymmetric fix-free codes. This result encourages the use of WSF codes to take advantage of the same forward and backward decoders and the low redundancy.

Automatic modulation recognition (AMR) of underwater acoustic (UWA) signals is incredibly challenging due to the complexity of UWA channels and the severity of ocean noise. In the presence of noise interference, single-modal features fail to fully represent the characteristics of different modulated signals. While the in-phase/quadrature (I/Q) and time-frequency maps can adequately represent the signal features in the time, frequency, and time-frequency domains, the direct integration of the two modalities is ineffective because of the variations in shape, information granularity, and noise manifestation. To address the low recognition rate caused by the above issues, we propose a two-stream transformer (TSTR) based network for AMR of UWA signals. First, the input pre-processing layer obtains the I/Q and time-frequency features from the received signals. Then, the feature capture layer extracts high-dimensional signal features in the time, frequency, and time-frequency domains. Finally, the classification layer estimates the modulation of the signals. A multi-head self-attention module with adaptive soft thresholding is used in the feature capture layer to provide noise reduction and redundant feature rejection while retaining context information. Moreover, multi-scale ghost convolution is employed to address the inability of the transformer to efficiently extract spatial characteristics from the signals. Results are presented using real UWA channels from the Watermark dataset for two different seas which show that the TSTR improves recognition by 1.2% and 5.9% over the best existing model. Further, it has better generalization capabilities and the model has a small number of parameters so the time complexity is low.

The number of mobile users and applications is rising quickly due to the deployment of fifth generation (5G) communication technology. The prevalence of smart devices and internet of things (IoT) services has made this technology essential to manage the resulting volume of data. However, the increase in mobile communications raises security concerns considering the open and dynamic nature of the industrial internet of things (IIoT) environment. An important research problem is how to exploit the characteristics of wireless channels for safe and reliable information transmission. Therefore, a mobile communication performance analysis and prediction algorithm based on FL-GLP-Net is proposed. First, a mobile security communication system model based on N-Nakagami channels is presented. Then, the non-zero secrecy capacity probability (NSCP) is studied and an exact expression is derived. XGBoost is used to choose the best features based on model performance. A real-time mobile security performance prediction model based on FL-GLP-Net is designed for real-time NSCP prediction. Federated learning (FL), graph attention network (GAT), long short-term memory (LSTM), and pyramid visual converter (PVT) modules are integrated to obtain a model that can deal with the diverse signal features in mobile communication systems. Results are presented to show that the proposed method outperforms other NSCP prediction algorithms. In particular, the mean squared error (MSE) of FL-GLP-Net is 78% better than that of FL-ShuffleNetV2.

Emerging technologies such as artificial intelligence and big data have made numerous Internet of things (IoT) applications possible. In particular, the Artificial Intelligence of Things (AIoT) has the potential to promote the digitization and intelligent connection of all things. However, the openness and diversity of AIoT makes data information vulnerable to security attacks which can lead to a disruption of mobile communication networks. The complexity of real-time data security events requires accurate prediction of AIoT security performance. In this paper, a secure communication system model based on decode-and-forward (DF) relaying is proposed and its security performance is analyzed. Expressions for the secrecy outage probability (SOP) are derived, and these are used to evaluate the security performance. For this purpose, an intelligent SOP prediction algorithm based on MS-Net is proposed. MobileNet and SqueezeNet networks are used to design an improved lightweight MS-Net model, which is composed of a depth separable convolution block and a fire module in parallel. The fire module is used to reduce the number of parameters in the first branch, and the depth-separable convolution block is employed in the second branch instead of the standard convolution. This can adapt to nonlinear characteristic in the AIoT safety data and reduce energy consumption. Afterwards, the convolutional block attention module(CBAM) attention mechanism is used to improve the model’s ability to capture features. The proposed algorithm provides better AIoT security performance than other algorithms. In particular, the mean squared error (MSE) is 68.1% better than that of RegNet.

Space information networks (SINs) have the potential to overcome the coverage and reliability limitations of cellular networks. As a result, SINs have received considerable attention as an enabling technology for sixth-generation (6G) networks to support global connectivity. SINs often employ a large number of low Earth orbit (LEO) satellites which are vulnerable to spoofing attacks. Thus, game theoretic physical layer authentication (PLA) based on Doppler frequency spread (DS) and received power (RP) attributes is proposed to provide effective authentication for these satellites. Hypothesis testing with a threshold is used to distinguish between legitimate and illegitimate (spoofer) satellites. Then, a zero-sum PLA game in which the ground station (GS) chooses the optimal detection threshold (τ*) to maximize its utility and a spoofing satellite (s) chooses the optimal attack probability (k*) to maximize its utility. Numerical results are presented to demonstrate the effectiveness of the proposed approach.

The rapid expansion of Internet of Vehicles (IoV) networks has facilitated high throughput and reliable vehicular communications. Mobile vehicular networks face the challenges: diversification of network equipment, user mobility, and the broadcast nature of wireless channels, so physical layer security modeling of IoV communication systems has become important. The complexity of wireless communication channels makes real-time prediction of secrecy performance challenging. This paper presents an analysis of secrecy performance for mobile vehicular networks. To ensure data secure transmission, we have employed the decode-and-forward (DF) relaying scheme. The signal-to-noise ratio (SNR) of the effective end-to-end link is employed to obtain the mathematical expression results, which can evaluate the secrecy performance. The theoretical secrecy performance is confirmed via simulation. Then, we design a dense-inception convolution neural network (DI-CNN) model, and propose a DI-CNN-based intelligent prediction algorithm.Transformer, ShuffleNetV2, RegNet and YOLOv5 methods are employed to analyze the performance of DI-CNN algorithm. It is shown that the DI-CNN approach has a prediction accuracy that is 48.8% better than Transformer.

Abstract Digital signature schemes are used for the authentication and verification of signatures. The Courtois–Finiasz–Sendrier (CFS) digital signature is a well‐known code‐based digital signature scheme based on the Niederreiter cryptosystem. However, it is not widely used due to the computation time of the signing algorithm. Most code‐based digital signature schemes are based on the Niederreiter cryptosystem. This paper proposes a new code‐based digital signature that is based on the McEliece cryptosystem. Key generation, signing, and verification algorithms are presented. The key generation algorithm constructs a public key using random inverse matrices. The signing algorithm has lower complexity and requires less computation time than the CFS scheme to sign a document. The verification algorithm is able to detect forgeries. It is shown that the proposed scheme is secure against public key structural attacks.

Internet of Things (IoT) devices have become ubiquitous due to the prevalence of the Internet. However, the open nature of wireless networks makes them susceptible to spoofing attacks. Moreover, their heterogeneous characteristics create authentication challenges which are increasing due to the tremendous growth in the number and variety of devices. Physical-layer authentication (PLA) provides a solution by utilizing the unique characteristics of wireless channels to aid upper layer authentication (ULA). In this article, an adaptive PLA scheme is proposed which exploits the antenna diversity inherent in multiple-input multiple-output (MIMO) systems. This scheme employs a one-class classification support vector machine (OCC-SVM) with the magnitude and real and imaginary parts of the received signals as features. Results are presented which show that this scheme provides robust authentication. The authentication performance is evaluated considering two majority voting schemes for IoT applications.

Multiobject tracking (MOT) technology can be deployed to the Internet of Things (IoT) devices to enhance the security and reliability of some video analysis applications, such as video surveillance and intelligent security system. However, since the IoT devices with limited computing capacity and storage, most existing MOT methods are difficult to deploy to IoT devices and exhibit poor tracking robustness in scenes with frequent occlusions, severe crowded, and scale variations. To alleviate the aforementioned issues, we propose a regression-based online MOT method. First, an object-aware embedding extraction module (OAEM) is designed to extract preliminary discriminative embedding of the object in the current frame. Then, an embedding aggregation module (EAM) is proposed to obtain high-quality aggregated embedding in temporal. Finally, the temporal embedding combined with the preliminary embedding extracted from the current frame to obtain a refined embedding feature for subsequent position prediction and association. Importantly, we achieve a beneficial interaction between embedding extraction, position prediction and association task. The proposed method does not suffer from significant memory consumption. Therefore, our method is a potential solution for intelligent video analysis on IoT devices. To evaluate the proposed method, we have conducted numerous experiments on the MOT16, MOT17, and MOT20 benchmark data sets. Results demonstrate that the proposed method can provide a more robust tracking performance compared to other optimal methods.

A new class of twinSudokutables (TSTs) is presented. These tables can be divided into both$s \times d$and$d \times s$subtables. They are constructed using the cyclotomic cosets of$Z_n$via two distinct vectors of cyclotomic coset elements and their Kronecker product. We prove that it is possible to generate$m$TSTs that are strongly mutually distinct (SMD), i.e., for every$0\leq i, j \leq m-1$, the$(i,j)$th entry of the tables contains different symbols. We also provide a method to construct$m$different TSTs that can be converted into twin solidSudokutables (TSSTs) as a perfect set of SMD TSSTs in order to make triplet solidSudokucubes (TSSCs). These TSSCs are symmetric cubes so that a cut from any of the six faces is a TSST. As a result, new twinSudokupuzzles (TSPs) and SMDTSPs are obtained that can be used to design new types ofSudokugames.

The heterogeneous characteristics of wireless mobile networks within the Internet of things (IoT) create authentication challenges due to the large number of devices with diverse requirements and capabilities. Physical layer authentication (PLA) can provide solutions for this heterogeneous environment using wireless channel attributes. In the paper, an adaptive lightweight physical layer authentication (ALPLA) scheme is proposed using machine learning (ML). Antenna diversity at the receiver is exploited to increase the number of features to achieve a high authentication rate (AR). A one-class classifier support vector machine (OCC-SVM) is used with the magnitude and real and imaginary parts of the received signal at each receive antenna as features. One-class classification is a ML technique for outlier and anomaly detection which uses only legitimate training data. The sounding reference signal (SRS) in the 5G uplink radio frame is employed to obtain the features. The proposed scheme is evaluated in an urban environment under different mobility conditions. Results are presented which show that this scheme provides a high AR with sufficient antenna diversity. Further it is superior to other approaches in the literature.

In the field of transportation, the Internet of Vehicles (IoV) is an important component of the Internet of Things. The vehicle-to-vehicle communication is particularly challenging in mobile IoV networks because they are operated in complex and highly variable environments. The mobile IoV transmission interruption level can be evaluated by the outage probability (OP) performance. If the OP performance can be analyzed and predicted accurately, the Quality of Service (QoS) in the mobile IoV networks can be improved. However, the analysis and prediction of mobile IoV transmission channels is very challenging because they are highly dynamic. In this article, the analysis and prediction of the OP performance for mobile IoV networks are investigated. A hybrid decode-amplify-forward (HDAF) relaying scheme with transmit antenna selection (TAS) is considered. The exact OP expressions are derived in a closed form, and the analytical results are verified. To realize the real-time analysis of the OP performance, an intelligent OP prediction algorithm based on the improved cuckoo search (ICS) is presented. The proposed algorithm is compared with different methods and the results show that it has a better OP prediction performance. The prediction accuracy of ICS-BP can be increased by 51.8% compared with the existing algorithms.

With the increasingly mature sensor technology and the increasing popularity of broadband network, “the Internet-of-Everything” era is coming, and the mobile Internet of Things (IoT) is booming around the world. However, the mobile IoT communication networks face serious challenges, which are caused by the complex and variable communication environments. The mobile IoT applications can produce large-scale data, which will consume substantial energy. The transmit antenna selection (TAS) and cooperative communication schemes are commonly used to reduce the complexity and the energy consumption, which directly impact the performance of mobile IoT networks. To evaluate the performance of mobile IoT networks, it is important to analyze outage probability (OP) performance. In this article, we investigate the OP performance analysis of mobile IoT communication networks and propose an OP intelligent prediction algorithm based on an improved convolutional neural network (CNN). First, the mobile OP performance is analyzed by combining the TAS and decode-and-forward cooperative schemes, and the exact OP expressions are derived. Then, an improved CNN is designed to avoid the loss of important information, which contains the input layer, three-convolution layer, one fully connected layer, and output layer. The proposed CNN-based prediction approach is compared with the radial basis function (RBF), generalized regression (GR), Elman, and extreme learning machine (ELM) methods. The simulation results validate that the proposed CNN prediction approach can achieve a better prediction effect than RBF, Elman, GR, and ELM methods. For the CNN approach, it has a 44% increase in the prediction accuracy.

Passive source positioning is of great interest due to the numerous applications. Energy based localization methods are popular because of their low cost and simplicity. In this paper, source localization with unknown transmit power is considered based on received signal strength difference (RSSD) measurements. An efficient three stage estimator is presented. First, a nonlinear RSSD-based model is formulated and the corresponding Fisher information for Gaussian distributed noise is derived. Next, a mean squared error (MSE) estimator is developed based on an unknown linear bias and Fisher information minimization. This results in a nonlinear optimization problem to minimize the MSE directly considering the linear bias. Finally, semidefinite relaxation is employed to transform this nonconvex problem into a convex minimization problem. This can be solved efficiently to obtain the optimal solution of the corresponding semidefinite programming (SDP) problem. Necessary and sufficient conditions for the optimality of the proposed RSSD based linear biased MSE SDP method (RLBM-SDP) are derived. Further, the corresponding bias sensitivity is formulated which yields an extension of the proposed method to a bounded RLBM-SDP (BRLBM-SDP) algorithm that includes a new constraint on the norm of the bias sensitivity. The computational complexity of the proposed methods is evaluated. Performance results are presented which confirm the efficiency of the proposed methods for sufficiently large signal to noise ratios.

Industrial applications can produce significant amounts of data that require low delay and high data rate communications. Multiple-input-multiple-output filter bank multicarrier (MIMO-FBMC) communications employing offset quadrature amplitude modulation has been proposed for industrial big data due to its reliability and high spectrum efficiency. One of the difficulties in implementing a MIMO-FBMC system is accurate channel estimation (CE). The main factor affecting the CE performance is intrinsic imaginary interference, and the conventional preamble-based CE is not effective in this case. Thus, in this article, a low-complexity sparse adaptive CE scheme is proposed that is based on a dynamic threshold. This reduces the number of inner product calculations by considering only the columns of the measurement matrix greater than the threshold. Simulation results are presented that show that the proposed scheme is better than other well-known methods in terms of computational complexity and CE accuracy.

A polar projection-based algorithm is proposed to reduce the computational complexity of dimensionality reduction in unsupervised learning algorithms. In particular, we consider the K-means clustering algorithm. A new distance metric is developed to account for peak power consumption to cluster consumer load profiles. This is used to cluster load profiles according to both total and peak power consumption. Numerical results are presented which demonstrate a significant reduction in computational complexity compared to K-means clustering using conventional dimension reduction techniques.

The extensive deployment of smart meters in millions of households provides a huge amount of individual electricity consumption data for demand side analysis at a fine granularity. Different from traditional aggregated system-level data, smart meter data is more irregular and unpredictable. As a result, probabilistic load forecasting (PLF), which can provide a better understanding of the uncertainty and volatility in future demand, is critical to constructing energy-efficient and reliable smart grids. In this article, a recently developed technique called Bayesian deep learning is employed to solve this challenging problem. In particular, a novel multitask PLF framework based on Bayesian deep learning is proposed to quantify the shared uncertainties across distinct customer groups while accounting for their differences. Further, a clustering-based pooling method is designed to increase the data diversity and volume for the framework. This not only addresses the problem of overfitting but also improves the predictive performance. Numerical results are presented which demonstrate that the proposed framework provides superior probabilistic forecasting accuracy over conventional methods.

Nonintrusive load monitoring (NILM) is a technique that infers appliance-level energy consumption patterns and operation state changes based on feeder power signals. With the availability of fine-grained electric load profiles, there has been increasing interest in using this approach for demand-side energy management in smart grids. NILM is a multilabel classification problem due to the simultaneous operation of multiple appliances. Recently, deep learning based techniques have been shown to be a promising approach to solving this problem, but annotating the huge volume of load profile data with multiple active appliances for learning is very challenging and impractical. In this article, a new semisupervised multilabel deep learning based framework is proposed to address this problem with the goal of mitigating the reliance on large labeled datasets. Specifically, a temporal convolutional neural network is used to automatically extract high-level load signatures for individual appliances. These signatures can be efficiently used to improve the feature representation capability of the framework. Case studies conducted on two open-access NILM datasets demonstrate the effectiveness and superiority of the proposed approach.

This paper considers power allocation in cellular networks over Rician fading channels. The goal is to improve the power consumption and energy efficiency as well as satisfy as many users as possible subject to user outage probability and normalized signal to interference plus noise ratio (SINR) constraints. The exact outage probability over Rician fading channels is determined using the moment-generating function (MGF). Further, upper and lower bounds on the outage probability are derived. These are used to characterize the relationship between outage probability and normalized SINR in Rician fading channels. Power allocation algorithms for power minimization and energy efficiency are proposed. Simulation results are presented to compare the performance of the proposed schemes with existing methods in terms of power consumption, throughput, energy efficiency, outage probability, and number of unsatisfied users.

Conventional interference suppression techniques based on scalar antenna array may not be effective, when the satellite and interference signals have similar directions of arrival. In this paper, a power inversion (PI) algorithm with a novel constraint is proposed which is applied to electromagnetic vector sensor (EMVS) arrays and improve the performance of the standard PI algorithm. Interference with different polarization parameters than the satellite signal can be suppressed even when they have the same incident direction. A fast implement scheme is designed for the proposed algorithm and is effective for the Beidou satellite system public service signal.

A dual hop, half‐duplex amplify and forward (AF) relay is considered for a source equipped with multiple transmit antennas. A power splitter is employed at the relay to harvest energy from the signal received from the source. The outage probability (OP) is derived considering optimal antenna selection to maximise the end‐to‐end signal‐to‐noise ratio. It is shown that the performance of the proposed energy harvesting based scheme improves with the number of transmit antennas. The effect of the power splitting factor on the OP is studied for different relay locations. Further, the OP is examined for different distances between the source and relay.

Power control is used in cellular communications systems to reduce power consumption and satisfy as many users as possible by managing the mutual interference between users. In signal to interference plus noise ratio (SINR) tracking power control (TPC) schemes, all users are required to adjust their power levels for each iteration, which is inefficient. In this study, a prioritised and selective uplink power control scheme is proposed. Priority user requirements are satisfied first, and then as many normal users (NUs) as possible are satisfied with their target SINRs. In addition, if an NU is currently satisfied with its target SINR, it is not required to update its transmit power level. Conversely, NUs who are not satisfied update their power levels. Simulation results are presented, which show that the proposed scheme outperforms dynamic target SINR TPC and variable target SINR TPC in terms of power consumption and efficiency. In addition, the proposed scheme is better than target SINR TPC and opportunistic power control as the number of users increases.

This study examines the linear source and relay precoder and destination combiner design for multiple‐input multiple‐output full‐duplex (FD) relay communication systems. The effect of the residual interference due to the imperfect loop interference cancellation is considered in the design. Two design algorithms are proposed to minimise the mean squared error of the received signal at the destination. The first is a tri‐step alternating iterative algorithm while the second is a bi‐step iterative algorithm, which has lower complexity and performance comparable to that of the first algorithm. The convergence of these algorithms is evaluated. Results are presented, which show that the proposed FD relay system can provide approximately double the achievable rate of the corresponding half‐duplex system if the residual interference is not high.

Tandem duplication (TD) errors occur when data is stored in the DNA of living organisms. The construction of codes to correct these errors was previously considered. A method was proposed to construct codes for TD errors of length at most k , , based on the uniqueness of ‐roots. It was shown that there exist words which have more than one ‐root when . As a consequence, the previous approach to correcting TD errors of length at most k cannot be extended to . In this study, ‐hinge‐free irreducible words are introduced and used to construct codes for TD errors of length at most k where . Furthermore, it is conjectured that the proposed approach can be extended to .

A linear network (LN) code can be described by either global encoding kernels (GEKs) or local encoding kernels (LEKs). In the literature, the multicast property of an LN code is described using GEKs, so the design algorithms for multicast LN codes employ GEKs to check this property. In this paper, a criterion is developed so that LEKs rather than GEKs can be used to check the multicast maximum distance separable (MDS) property of linear network error correction (LNEC) codes on erroneous networks. Further, it is used to develop a design algorithm for multicast MDS LNEC codes on erroneous cyclic networks. This algorithm is more efficient than the algorithm that uses GEKs when the number of edges is high and the interconnection of these edges is low.

Passive source localization is an important research area with numerous applications in mobile communications and signal processing. This paper presents an analysis of energy-based localization performance with unknown transmit power using received signal strength (RSS) and RSS difference (RSSD) models. These models are widely used because of their low cost and simple implementation. Previous studies were based on the assumption that the source node transmit power is known, which is not practical in many situations. The Cramér-Rao lower bound for the RSS and RSSD models with correlated noise is derived as a performance benchmark for the mean squared error (MSE) of the location and transmit power estimation. It is shown that the MSE can be factored into two independent terms corresponding to the geometric distribution of the sensors and the channel parameters including the noise variance. The effect of the sensor and source node positions on the location accuracy and the MSE is derived via the geometric dilution of precision (GDOP). The GDOP is then used to evaluate the effect of the joint estimation of unknown power and source location on the performance. Lower bounds on the GDOP are derived to obtain the minimum MSE. The RSSD solution is obtained by formulating the nonlinear and nonconvex objective functions into a convex optimization problem through relaxation and semidefinite programming. Simulation results are presented with the source inside or outside the sensor network.

LZ77, LZ78 and other compression algorithms use derivates of the original Lempel-Ziv (LZ) parsing algorithm, which computes the LZ production complexity. The focus of this paper is on the original algorithm. Like its derivates, it progressively factors an input string s into a series of steps, each of which consists of a reference to a substring starting in the already parsed part of the s (reproduction) and a single symbol (innovation). In all LZ parsers, the input string generally ends on an incomplete step part-way through a reproduction rather than on an innovation symbol. This LZ parsing tail is usually given scant treatment as it tends to be short compared to the string length |s|, even though its length is bound only by |s|- 1. This paper examines the expected length of the tail for |s| produced by an i.i.d. Bernoulli source, and uses this to derive a new result on the statistics of subsequences of large strings.

The following topics are dealt with: public domain software; software maintenance; Java; program diagnostics; learning (artificial intelligence); program compilers; C language; software reusability; source code (software); software engineering.

Millimetre‐wave (MMW) in 60 GHz extremely high‐frequency band is deemed as a promising technique in indoor localisation based on its high multipath resolution, which can achieve higher measurement precision than ultra‐wideband system. To reduce the positioning errors, this study presents a new threshold determination scheme based on the extreme learning machine by analysing the characteristics of the received MMW pulses from the energy detector receiver. For each given signal‐to‐noise ratio, the thresholds are examined. The impacts on ranging accuracy are analysed due to the integration periods and propagation models. Results show the developed method can extensively improve the ranging precision.

Adaptive modulation is widely employed to improve spectral efficiency. To date, square signal constellations have been used with adaptive modulation. In this paper, triangular constellations are considered for this purpose. Triangle quadrature amplitude modulation (TQAM) for both power-of-two and non-power-of-two modulation orders is examined. A technique for TQAM mapping is presented which is better than existing approaches. A new type of TQAM called semi-regular TQAM (S-TQAM) is introduced. Bit error rate expressions for TQAM are derived, and the detection complexity of S-TQAM is compared with that of regular TQAM (R-TQAM) and irregular TQAM (I-TQAM). The performance of S-TQAM over additive white Gaussian noise and Rayleigh fading channels is compared with that of R-TQAM and I-TQAM.

In this paper, a new symmetric key cryptosystem based on linear codes is presented. Unlike most code‐based cryptosystems, the code is known publicly. The encryption is based on randomly interleaving two codewords containing random errors. This is the first code‐based cryptosystem based on random interleaving. The key size and security of the proposed scheme are analysed, and it is shown that allowing the code to be public results in a significantly smaller key than other code‐based cryptosystems. Furthermore, it is proven that the best possible chosen plaintext attack does not affect the security.

Device‐to‐device (D2D) communications underlaying a cellular network is an efficient way to enhance spectral efficiency via resource sharing between D2D and cellular users (CUs). In this study, a discrete location‐aware (DLA) interference model for D2D users is presented to allocate cellular resources. The vicinity of a CU is discretised into multiple regions, and the number of active D2D users in each region is constrained to satisfy the CU QoS requirements. Considering the locations of D2D users affecting the interference to CUs and thus the achievable rate, the formulated non‐linear 0–1 knapsack resource allocation (RA) problem is divided into two subproblems: (i) the optimal amount of shared resources between the two types of users; (ii) the optimum subset of D2D users which transmit . The conditions of D2D users spatial deployment and resources reuse portion to achieve the solutions of the two subproblems are theoretically derived and proven. Then a DLA‐RA algorithm is proposed to solve the corresponding subproblems in both single CU and multiple CUs cases. Extensive simulations results are presented which verify the effectiveness of the proposed DLA interference model and the RA scheme.

Extreme learning machine (ELM) has attracted considerable attention in recent years due to its numerous applications in classification and regression. In this study, the authors investigate the performance of an ELM‐based threshold selection algorithm for 60 GHz millimetre wave time of arrival estimation using energy detector (ED). A hybrid metric based on the skewness, kurtosis, standard deviation, and slope of the ED values is employed. The optimal normalised threshold for different signal‐to‐noise ratios (SNRs) is investigated, and the effects of the integration period and channel model are examined. Performance results are presented which show that the proposed ELM‐based algorithm provides high precision and better robustness than existing techniques over a wide range of SNRs for the IEEE 802.15.3c CM1.1 and CM2.1 channel models. Further, the performance is largely independent of the integration period and channel model.

Traditional cellular networks require the downlink (DL) and uplink (UL) of mobile users (MUs) to be associated with a single base station (BS). However, the power gap between BSs and MUs in different transmission environments results in the BS with the strongest downlink differing from the BS with the strongest uplink. In addition, the significant increase in the number of wireless machine type communication (MTC) devices accessing cellular networks has created a DL/UL traffic imbalance with higher traffic volume on the uplink. In this paper, a joint user association and resource partition framework for downlink-uplink decoupling (DUDe) is developed for a tiered heterogeneous cellular network (HCN). Different from the traditional association rules such as maximal received power and range extension, a coalition game based scheme is proposed for the optimal user association with DUDe. The stability and convergence of this scheme are proven and shown to converge to a Nash equilibrium at a geometric rate. Moreover, the DL and UL optimal bandwidth partition for BSs is derived based on user association considering fairness. Extensive simulation results demonstrate the effectiveness of the proposed scheme, which enhances the sum rate compared with other user association strategies.

Recently, the refined singleton bound over acyclic networks was extended to convolutional error-correcting network codes over cyclic networks using extended coding vectors. In this paper, it is shown that constructing an MDS code is equivalent to constructing a multicast code. This is used to develop an algorithm for constructing MDS field-based codes over acyclic networks when the sinks know the topology of the network and the network coding employed at all nodes. A lower bound is given on the size of the field required for the algorithm to be successful. Then this algorithm is extended to construct MDS convolutional error-correcting codes over cyclic networks. The complexity of the proposed algorithm is evaluated.

Cellular technology has the potential to support large numbers of machine-type communications (MTC) devices for a variety of applications in fifth generation wireless systems. As MTC devices are a recent addition to cellular networks, a major concern is how to effectively manage and limit cellular resources for MTC data transmission without degrading traditional human-type communications (HTC) performance. To tackle this problem, a two-stage resource management framework is proposed with the goal of maintaining traffic equilibrium. In the first stage, an ecological prey-predator model is introduced to model the resource partition for the two types of devices. The steady-state properties of the traffic are analyzed, and the value regions of the allocated MTC resources for stable equilibrium points are derived. In the second stage, given an appropriate resource partition for MTC traffic, these devices are grouped based on their buffer conditions. The optimal resource allocation solution is derived so that the MTC traffic is stable. It is shown that to maintain MTC traffic stability, the resources can be allocated to only two groups. Furthermore, results are presented which show that HTC and MTC traffic can maintain a stable equilibrium using the two-stage resource management framework.

Pedestrian detection systems are receiving increasing attention in both industry and academia with the rapid development of autonomous automobiles which employ artificial intelligence. These systems must detect specific classes of objects such as pedestrians rather than generic objects. In this paper, we present a faster RCNN based pedestrian detection system which improves upon previous solutions. The proposed model takes arbitrary size images as inputs and generates bounding boxes and confidence scores for pedestrians. The system achieves good performance and is faster than the well known and frequently used methods in the literature.

A generalised selection transmission (GST) and generalised selection combining (GSC) scheme is proposed for two‐way relay networks where two multi‐antenna sources exchange information via a single‐antenna relay. New exact and asymptotic expressions are derived for the outage probability and symbol error rate (SER) in Rayleigh fading. Moreover, a tight upper bound on the ergodic sum‐rate is presented. These results are used to demonstrate that the proposed GST/GSC scheme preserves the full diversity order, which equals the minimum number of antennas at the two sources. It is also shown that the impact of the number of selected antennas lies in the array gain only. Furthermore, the GST/GSC scheme significantly improves the performance relative to single‐antenna selection, and only incurs a negligible reduction in performance relative to all‐antenna beamforming. Finally, the optimal relay location that minimises the SER is determined analytically. It is observed that the optimal relay location shifts towards one source when the number of selected or available antennas at the other source increases.

The compressed spectrum sensing problem for cognitive radio (CR) spectrum sharing in device-to-device (D2D) communications is investigated, with an emphasis on collaborative sensing strategies for multi-user CR-based D2D networks. Because the D2D users are assumed to be geographically close to each other, the same spectral occupancy is expected, which can be exploited in the sensing algorithm design. We first investigate a single user compressed spectrum sensing algorithm, where the successive fast iterative shrinkage-thresholding algorithm (FISTA) is employed. The successive FISTA-based single-user sensing algorithm is then used to develop two collaborative compressed spectrum sensing schemes, namely Equal-Gain Combining (EGC) and Census-Weighted Detection Results Combining (CWDRC). It is demonstrated that both algorithms are effective in low signal-to-noise ratio (SNR) conditions. We further show that the census-based CWDRC algorithm significantly reduces system overhead compared to other collaborative sensing strategies with only a small degradation in the probability of detection. Thus, CWDRC is attractive for collaborative spectrum sensing in CR-based D2D systems.

In this paper, a new secure channel coding scheme is presented which randomly inserts and deletes bits in a codeword of a quasi‐cyclic‐low‐density parity check (QC‐LDPC) code. It is shown that the key size is smaller than other code‐based cryptosystems based on permutation and scrambling matrices. The positions of the inserted and deleted bits are determined using a secret key. It is shown that the error performance of the resulting code after the insertions and deletions is better than a random low‐density parity check code with similar parameters. An important advantage of this cryptosystem is that even if the QC‐LDPC code is revealed, the system remains secure. Furthermore, the proposed approach using insertions and deletions can be employed with other classes of error correcting codes.

This study considers the construction of four‐cycle free quasi‐cyclic low‐density parity‐check (QC‐LDPC) codes. These codes are based on the cyclotomic cosets of q modulo n where q is a prime power, n is a prime and ( n , q ) = 1. If s is the order of q modulo n , then n − 1 = sl for some integer l . Then there are l distinct cyclotomic cosets , 1 ≤ i ≤ l . Concatenation of the l circulant matrices formed using these sets gives an s × sl matrix M co ( n , q ). In addition, the Kronecker product of the transpose of ∪ i with itself modulo n gives an sl × sl matrix M kr ( n , q ). Replacing the entries of these matrices with their associated dispersed binary n × n matrices provides binary QC‐LDPC codes with girth at least six. Furthermore, for any prime power q ′ such that q ′ = n + 1 or q ′ ≥ 2 n , these two matrices can be used to construct q ′‐ary QC‐LDPC codes with girth at least six. These constructions produce large classes of four‐cycle free binary and non‐binary QC‐LDPC codes. Among the structured LDPC codes, the introduced technique, from code‐construction‐complexity perspective, which is an engineering factor, is at least among the best ones if not the best. Performance results are presented which show that the codes obtained perform well over additive white Gaussian noise channel with the iterative sum‐product decoding algorithm.

The heterogeneous smart grid (SG) poses two major challenges for wireless networks, namely, providing sufficient bandwidth for a wide variety of applications and high reliability for critical real-time applications. To address these challenges, the impact of communication outage on the demand response management as a typical SG application is analyzed in this paper. A dynamic spectrum management (DSM) technique is proposed to allocate resources, considering the QoS and application priorities. Vacant digital TV frequency bands are utilized to support SG applications. An algorithm to estimate the SG capacity is introduced, which can be applied to various user distributions and SG environments. This is used in conjunction with a low-complexity coloring theory algorithm to allocate the spectrum. The results presented show that DSM provides better performance than traditional fixed spectrum management, in terms of QoS and secondary spectrum utilization.

Source localization is an important field of research with numerous applications in signal processing and wireless communications. In this paper, we present a received signal strength difference (RSSD) based method to localize a source with unknown transmit power in the presence of sensor position uncertainty. It is an efficient low complexity technique which does not require transmit power estimation as with other approaches. This eliminates the uncertainty due to signal propagation parameter variations. A constrained adaptive weighted least squares technique is presented to obtain a least squares initial estimate (LSIE) of the source location. Then, this estimate is improved using a computationally efficient modified Newton method (MNM) with adaptive weights. The bias of the proposed LSIE-MNM method and the Cramér-Rao lower bound (CRLB) of the RSSD based measurement model are derived to determine the effect of sensor position uncertainties on the source location estimate. Results are presented which show that the proposed method achieves the CRLB when the SNR is sufficiently high.

Queue-aware transmission scheduling for cooperative wireless communications with sub-fading-block scheduling to better balance load and capacity in low mobility environments is investigated. The scheduling problem for joint cooperation scheduling and resource allocation is formulated as a constrained nonlinear integer optimization problem over an integer convex set based on a source buffer queueing analysis. It is shown that with queue-aware scheduling, the state transition matrix of the source buffer queue has a highly dynamic form. As a result, the objective function of the optimization problem does not have an analytic form in general. The constrained discrete Rosenbrock search algorithm, which is a gradient-free directed discrete search algorithm, is employed to solve the nonlinear integer problem. The output of the directed integer search algorithm is used for queue-aware transmission scheduling for the cooperative system. Numerical results are presented which show that, for cooperative transmission scheduling, the Rosenbrock search based queue-aware algorithm significantly outperforms the equal partitioning, random partitioning, and gradient-based algorithms under quasi-static channel assumptions. Under practical system conditions with unsaturated traffic, the proposed queue-aware scheduling scheme achieves the true optima, and maintains a large stability region for the buffer queue, over a wide range of channel and traffic conditions. It is also shown that when fading channel dynamics are taken into consideration, the performance of the proposed queue-aware scheduling algorithm significantly outperforms fixed relaying and fixed direct transmission channel-aware scheduling strategies.

This paper presents a practical key generation algorithm based on the reciprocity of wireless fading channels. A broadband chaotic signal is employed for transmission so that the fading is frequency selective. In this case, signal components in the frequency domain spaced greater than the coherence bandwidth of the channel can be considered uncorrelated. The proposed algorithm exploits this property to generate a unique shared key between two parties. The nonperiodicity of the chaotic signal provides a unique signal for key generation, which can be used even with static fading channels. The proposed approach is robust to timing differences between the parties because the frequency spectrum of the signals is employed. A technique for information reconciliation is presented which does not reveal any information about the values used to generate the key. The randomness of the key is confirmed, and the effects of additive white Gaussian noise and timing differences on the performance of the algorithm are examined.

In the context of fix-free codes, the most important and immediate consequence of the 3/4-conjecture (if it is proven), is that the redundancy of an optimal binary fix-free code never exceeds 1 bit, as with the optimal prefix-free codes, i.e. Huffman codes. In this paper, this bound on the redundancy is proven without requiring the conjecture to be true. To do so, we use two known sufficient conditions for the existence of binary fix-free codes to derive an improved upper bound on the redundancy of an optimal fix-free code in terms of the largest symbol probability.

A mobile social network (MSN) is defined as a mobile network that uses social relationships to determine node communication. There has been extensive research on routing protocols for MSNs. These protocols assume cooperating nodes carry messages to the destination nodes. However, since these protocols employ social relationships to facilitate communication between nodes, node cooperation may be limited due to privacy concerns. Thus, a privacy strategy is proposed here for MSN routing protocols. Performance results are presented which show that this approach leads to greater node cooperation and thus a better probability of delivery.

In this paper, the application of radio over fiber (RoF) based wireless communications for smart metering and power distribution is introduced. A RoF based network has the potential to enhance both the coverage and spectral efficiency of smart grid wireless networks since its major benefit is adaptability to temporal and spatial traffic variations. A clustered architecture is presented which provides signal diversity to enhance the coverage and capacity of smart grid communications. Examples are presented to illustrate the improvement in coverage and latency of a smart grid wireless network when RoF is used instead of a conventional wireless network.

Compressive sensing (CS) has recently been proposed to recover nonlinearly distorted orthogonal frequency division multiplexing (OFDM) signals. This approach is only effective when the nonlinear distortion is very sparse, and as a result the power amplifier (PA) needs to have a large linear range. However, practical PAs typically have a small linear range to improve power efficiency and reduce cost. In this case, the nonlinear distortion has poor sparsity, which results in large estimation errors with CS-based methods. Further, the out-of-band radiation caused by nonlinear PA distortion cannot be mitigated since CS-based techniques are performed at the receiver. To address these problems, phase rotated pilot signals are introduced to modify the OFDM signal at the transmitter to ensure that the clipping noise is sufficiently sparse. This improves the performance of CS-based methods for PAs with a small linear range. In addition, the proposed technique reduces the peak-to-average power ratio (PAPR) and thus also the out-of-band radiation.

A locally decodable source code (LDSC) allows the recovery of arbitrary parts of an unencoded message from its encoded version, using only a part of the encoded message as input, a challenge that arises when searching within compressed data sets. Simple source codes such as Huffman codes or Lempel-Ziv compression are not well suited to this task: A decoder starting at an arbitrary point within the compressed sequence generally cannot determine its position with respect to the boundaries between encoded symbols, or requires information found before the starting point in order to be able to decode. In this paper, we propose the use of subsets of self-synchronising variable-length T-codes as source codes and show that local decoding is feasible and practical using subsets of T-codes with bounded synchronisation delay (BSD).

An intermittently connected network (ICN) is a mobile network that uses cooperation between nodes to facilitate communication. This cooperation consists of nodes carrying messages from other nodes in order to deliver it to the destination. An ICN does not require an infrastructure, and it does not store routing information, so it is an attractive networking environment. However, the challenge is how to achieve a high probability of message delivery with no routing information or network infrastructure. In this paper, network coding is employed to improve the delivery probability in an ICN. The benefit of using network coding over message replication is determined analytically and verified by simulation.

Passive source localization is an important area of research. In this paper, a received signal strength difference (RSSD) based localization technique is presented for wireless sensor networks. The advantage of this solution is that the transmit power does not have to be known. It is an effective low complexity technique as it eliminates the transmit power estimation required with other approaches. The uncertainties due to the signal propagation parameters are eliminated by using the received signal strength differences from known sensors. This yields a set of equations using signals from the source to sensors with known positions assuming a log-normal path loss model. The constrained weighted least squares (CLS) technique is used for location estimation using these equations. Results are presented which show that the performance of the proposed method achieves the Cramer-Rao lower bound (CRLB) for a sufficiently high signal to noise ratio (SNR).

Passive source location determination is a very active research area. In this paper, we present a received signal strength difference (RSSD) source localization method based on a total least square (TLS) estimator. Due to errors in the data vector and system matrix, the least squares (LS) and weighted least squares (WLS) methods are not applicable as they produce large bias in the location estimation. Therefore, an extension of the LS methods, called total least squares (TLS) is used to solve this problem. Due to the relationship between the data vector and system matrix, a modified TLS method is presented to achieve a closed form estimate. The advantage of this approach is that it does not require transmit power estimation as with other methods, and the complexity is low. The received signal strength difference is used to eliminate uncertainties due to the signal propagation parameters. Performance results are presented which show that the performance of the proposed method comes close to achieving the Cram'er-Rao lower bound.

Full data exchange with complex field network coding (CFNC) has been shown to achieve a throughput of 1/2 symbol per user per channel use (sym/U/CU) in a multiway relay channel (MWRC). Further, optimum precoding has been designed for a MWRC with a throughput of 1/2 sym/U/CU such that a rectangular quadrature amplitude modulation (QAM) constellation symbol is received. To achieve this throughput, any superimposed combination of user symbols must be distinguishable at the relay and users. In this paper, a decode-and-forward (DF) transmission scheme for full data exchange in a MWRC is presented which decreases the constellation size received by the users and hence leads to a downlink performance improvement. This is achieved by exploiting user self-information. It is also shown that the proposed DF transmission scheme leads to uplink performance improvement. The proposed transmission scheme is evaluated in both additive white Gaussian noise (AWGN) and Rayleigh fading channels.

In this study, a method is presented to construct column weight 3 (CW3) low‐density parity‐check (LDPC) codes using three‐partite graphs. Let G b be a bipartite graph and N g be the set of all minimum length cycles in G b . Using G b and N g , a three‐partite graph denoted G ( G b , N g ), or simply G t , is formed. Let T be the set of length 3 cycles in G t and T a be the set of three element subsets of vertices in G t such that each of these subsets form a subgraph with no edges in G t and has precisely one element in each section of G t . Furthermore, let H be the binary matrix in which the set of rows represent the set of vertices of G t , the columns represent the elements of V := T ∪ T a , and h ij = 1 if and only if the i th vertex of G t belongs to the j th three element set in V . Then H is a CW3 binary matrix. Using the Tanner graph representing H , a recursive construction for CW3 LDPC codes is provided. Applying a simple restriction on T and T a , codes free of length 4 cycles are generated. Euclidean and finite geometry codes are used as the base codes for generating new CW3 LDPC codes. Results are presented which show that these new codes perform well in an additive white Gaussian noise (AWGN) channel with the iterative sum‐product decoding algorithm.

A new joint cryptography‐channel coding technique is introduced which employs punctured quasi‐cyclic‐low density parity check (QC‐LDPC) codes obtained from extended difference families. The absence of permutation and scrambling matrices reduces the key size compared with similar code‐based cryptosystems, while having an acceptable level of security. The main advantage of this system is that, provided the system parameters are chosen appropriately, even if the code employed is revealed the system remains secure. Performance results are presented which show that the punctured code outperforms a random low density parity check (LDPC) code of the same length and rate.

This paper presents flott, a fast, low memory T-transform algorithm which can be used to compute the string complexity measure T-complexity. The algorithm uses approximately one third of the memory of its predecessor while reducing the running time by about 20 percent. The flott implementation has the same worst-case memory requirements as state of the art suffix tree construction algorithms. A suffix tree can be used to efficiently compute the Lempel-Ziv production complexity, which is another measure of string complexity. The C-implementation of flott is available as Open Source software.

In this paper, we investigate the design of a practical information-theoretically secure secret key agreement protocol for a cooperative wireless network employing standard modulation. Assuming relay selection has been completed, the key agreement problem is studied in a three-node cooperative wireless communication system over block-fading channels. Passive attacks from an eavesdropper collocated with the relay are considered. We derive upper and lower bounds on the secret key rate of this cooperative wireless system. The difference between the bounds is shown to be small for practical communication scenarios, which indicates they are tight. We then propose a practical secret key agreement protocol for this system with both the communicants and the honest relay participating in the public discussion. The tradeoff between security and protocol efficiency is considered in the joint design of advantage distillation, information reconciliation, and privacy amplification. The protocol parameters are optimized to achieve the tight bound on the secret key rate.

Complex field network coding (CFNC) has been shown to achieve a throughput of 1/2 symbol per user per channel use in a multi-way relay channel (MWRC). To achieve this throughput, a superimposed combination of user symbols must be distinguishable at the relay. In this paper, full data exchange in a MWRC network with fading is considered. The pairwise error probability (PEP) of a MWRC with Rayleigh fading is presented, and a closed form approximation of the minimum Euclidean distance distribution for the relay constellation is given. Tight upper bounds on the symbol error rate (SER) are obtained using a nearest neighbours approximation and the effect of a precoding vector on system performance is investigated.

Throughput scaling laws for two coexisting ad hoc networks with m primary users (PUs) and n secondary users (SUs) randomly distributed in an unit area has been widely studied. Early work showed that the secondary network performs as well as stand-alone networks, namely, the per-node throughput of the secondary networks is equation. In this paper, we show that by exploiting directional spectrum opportunities in secondary networks, the SU throughput can be improved. If the main lobe of the SU antenna pattern can be as narrow as possible, then the SUs can achieve a per-node throughput of equation which is Θ(log n) times higher than the the throughput without directional transmission. If we consider practical constraints and assume the minimum angle of the main lobe is δth, then the SU throughput gain is equation compared with the throughput without directional transmission. We also explore the statistics of directional spectrum holes in this paper.

Sperm whales (Physeter macrocephalus) emit acoustic signals when diving to search for food. Because they dive to depths of over 1500 m, conventional localization methods are impractical. Sperm whales produce a series of loud impulsive echolocation sounds called `clicks' at a fairly constant rate, and these signals can be used for localization. A geometric approach is considered using signals from the direct and surface reflection paths. The time difference of arrival (TDOA) between these paths is used for localization. Real sperm whale data from the Atlantic Undersea Test Center (AUTC) is used to evaluate the proposed method.

Reverberation is a major problem for time delay estimation (TDE) in enclosed environments. In this paper, a robust TDE method based on the generalized cross-correlation (GCC) is proposed. An efficient preprocessing technique to calculate the all-pass component is introduced to improve the performance of GCC-based techniques in reverberant conditions. Performance results are given which demonstrate that the proposed approach provides better performance for a wide range of microphone locations and reverberation times. Results in real acoustic environments confirm the effectiveness of the proposed TDE method. Compared to other TDE methods, our solution has low computational complexity and can be employed in real-time applications.

This paper considers a heterogeneous communication network consisting of a Smart Grid Neighborhood Area Network (SG-NAN) and a Digital Video Broadcasting (DVB) system. A Dynamic Spectrum Management (DSM) technique is proposed for this network to allocate resources to the SG-NAN and DVB systems to satisfy SG-NAN QoS requirements while limiting interference to the DVB system. A novel algorithm for estimating SG-NAN capacity is introduced which can be applied to various user distributions and hybrid service situations. The DVB capacity is also calculated under interference constraints. Based on these capacities, spectrum is allocated using a coloring theory algorithm to reduce the complexity. Simulation results are presented which show that DSM provides better performance than traditional fixed spectrum management (FSM) in terms of QoS and secondary spectrum utilization.

This paper considers cyclic DNA codes of arbitrary length over the ring R = F2[u]/(u4- 1). A mapping is given between the elements of R and the alphabet {A, C, G, T} which allows the additive stem distance to be extended to this ring. Then, cyclic codes over R are designed such that their images under the mapping are also cyclic or quasi-cyclic of index 2 with designed hybridization energy. The hybridization energy and additive distance are shown to be functions of the neighborhood energy.

Efficient spectrum utilization is of great importance in cognitive networks, however current spectrum sharing techniques in the temporal or spatial domains all have deficiencies. In this paper, we propose a joint spatial and temporal spectrum sharing scheme based on the concept of regions. We define four regions, namely a primary exclusive region (PER), a temporal spectrum sharing region (T-SSR), a joint spectrum sharing region (J-SSR), and a spatial spectrum sharing region (S-SSR). TSSR and J-SSR are proposed to utilize temporal spectrum holes which have not previously been exploited. Cooperative spectrum sensing is employed to expand the J-SSR, which is beneficial for system design. Closed-form bounds for the four regions are obtained, and conditions on T-SSR existence and the presence of a transition zone between the JSSR and S-SSR are determined. Both analytical and simulation results are presented which show how the key factors, including primary user interference constraints, spectrum sensing factors, and secondary user coverage constraints, influence these bounds. This has great practical value in improving spectrum efficiency in cognitive networks.

In this paper, we investigate the connectivity of random cognitive radio networks with different routing schemes. Two coexisting ad hoc networks are considered with m primary users (PUs) and n secondary users (SUs) randomly distributed in a unit area. The relation between n and m is assumed to be n = mβ. We show that with the HDP-VDP routing scheme, which is widely employed in the analysis of throughput scaling laws of ad hoc networks, the connectivity of a single SU can be guaranteed when β > 1, and the connectivity of a single secondary path can be guaranteed when β > 2. While circumventing routing can improve the connectivity of cognitive radio ad hoc network (CRAHN), we verify that the connectivity of a single SU as well as a single secondary path can be guaranteed when β > 1. Thus to achieve the connectivity of secondary networks, the density of SUs should be larger (asymptotically) than that of the PUs. A smart routing scheme can also improve the connectivity of CRAHN. Our results serve as a guide to deployment and routing design for cognitive radio networks.

The Radio Environment Map (REM) provides an effective approach to Dynamic Spectrum Access (DSA) in Cognitive Radio Networks (CRNs). Previous results on REM construction show that there exists a tradeoff between the number of measurements (sensors) and REM accuracy. In this paper, we analyze this tradeoff and determine that the REM error is a decreasing and convex function of the number of measurements (sensors). The concept of geographic entropy is introduced to quantify this relationship. And the influence of sensor deployment on REM accuracy is examined using information theory techniques. The results obtained in this paper are applicable not only for the REM, but also for wireless sensor network deployment.

Pulse waveforms for 60 GHz impulse radio (IR) communication systems are investigated. The power spectral densities of the pulses are examined for compliance with Federal Communications Commission (FCC) spectral regulations. The error probability and capacity of M ‐ary pulse position modulation (PPM) 60 GHz IR systems with a correlation receiver using different pulse waveforms are analysed. Both orthogonal and non‐orthogonal PPM is considered with additive white Gaussian noise and IEEE 802.15.3c channel models. Analytical and simulation results are presented, which show that an autocorrelation function that decays quickly provides a lower probability of error and better capacity. Among the pulse waveforms examined in this study, the truncated sinc pulse is the best in terms of compliance with FCC regulations and the PPM error probability and capacity.

This paper presents a single-microphone approach to the enhancement of noisy reverberant speech via inverse filtering and spectral processing. An efficient algorithm is used to blindly estimate the inverse filter of the Room Impulse Response (RIR). This filter is used to attenuate the early reverberation. A simple technique to blindly determine the filter length is presented. A two-step spectral subtraction method is proposed to efficiently reduce the effects of background noise and the residual reverberation on the equalized impulse response. In general, the equalized impulse response has two detrimental effects, late impulses and pre-echoes. For the late impulses, an efficient spectral subtraction algorithm is developed which introduces only minor musical noise. Then a new algorithm is introduced which reduces the remaining pre-echo effects. The performance of this two-stage method is examined in different reverberant conditions including real environments. It is also evaluated with white Gaussian and recorded babble noise. The results obtained demonstrate that the proposed blind method is superior in terms of reducing early and late reverberation effects and noise compared to well known single-microphone techniques in the literature.

This paper presents a novel iterative hard decision decoding algorithm for binary linear block codes over a binary symmetric channel (BSC). The problem is formulated as a 0-1 integer programming problem which is known to be NP-hard. When the crossover probability c of the channel is known, the solution space of the decoding problem can be decreased to a sphere whose radius is related to c. Using the penalty function method, the problem is reformulated on this reduced solution space. Then an iterative multi-flip local search algorithm is designed to find the global solution of this decoding problem. For a code with minimum distance d, when the radius of the sphere is not greater than d-1/2, this algorithm has the maximum likelihood (ML) certificate property, i.e., if the decoder outputs a codeword, it is guaranteed to be the ML codeword. Compared to the probabilistic suboptimal iterative belief propagation (BP) decoder, this approach has lower complexity and better performance. Numerical results show that in terms of speed and performance the proposed decoding method outperforms BP decoding in the error floor region.

In order to determine how suboptimal the Shannon code is, one should compare its performance with that of the optimal code, i.e., the corresponding Huffman code, in some sense. It is well known that in the worst case the redundancy of both the Shannon and Huffman codes can be arbitrarily close to 1. Beyond this worst case viewpoint, very little is known. In this paper, we compare the performance of these codes from an average point of view. The redundancy is considered as a random variable on the set of all sources with n symbols and its average is evaluated. It is shown that the average redundancy of the Shannon code is very close to 0.5 bits, whereas the average redundancy of the Huffman code is less than n-1(1+ln n)+0.086 bits . It is also proven that the variance of the redundancy of the Shannon code tends to zero as n increases. Therefore, for sources with alphabet size n, the redundancy of the Shannon code is approximately 0.5 bits with probability approaching 1 as n→ ∞.

We study practical information-theoretically secure secret key agreement protocol design for wireless cooperative communication networks. By assuming the relay selection process is finished in advance, we study the key agreement problem for a classical three-node cooperative wireless communication system over independent additive white Gaussian noise (AWGN) channels. Passive attacks from an eavesdropper collocated with the relay are assumed. We derive upper and lower bounds on the secret key rate under the assumed cooperative wireless system model and find that they are close, which indicates the tightness of the bounds. We then propose an efficient practical secret key agreement protocol with both Bob and the honest relay participating in the public discussion. A compromise between security and efficiency is achieved by the joint advantage distillation and privacy amplification.

In this paper, we consider the high peak-to-average power ratio (PAPR) problem of non-contiguous orthogonal frequency division multiplexing (NC-OFDM) signals in cognitive radio systems. A high PAPR can lead to saturation in the power amplifier (PA) of secondary users (SUs) and consequently increase spectral spreading, and cause interference to adjacent primary users (PUs). To overcome this problem, existing PAPR reduction techniques for OFDM systems can be applied to NC-OFDM, but they should provide a low PAPR with no side information and relatively low complexity. We consider NC-OFDM with blockwise-subcarrier activation and show that it can intrinsically employ tone reservation (TR) as a PAPR reduction technique. The proposed TR reserves subcarriers within inactive subblocks that are not used by the primary and secondary users. This eliminates data rate loss due to reserved peak reduction tones (PRTs). Further, dynamic PRT allocation in NC-OFDM typically requires side information about the PRT locations to be sent to the receiver. Since we choose PRTs from inactive subblocks, they are simply discarded at the receiver without any side information. The power spectral density (PSD) and bit error rate (BER) are evaluated at the output of the nonlinear PAs to provide a realistic performance comparison.

Tone reservation (TR) has been proposed to reduce the peak-to-average power ratio (PAPR) of orthogonal frequency division multiple access (OFDMA) uplink communication systems. However, it introduces multiple-access interference (MAI) at the receiver. To reduce the effects of MAI, TR was proposed which uses all unused subcarriers at the OFDMA transmitter to generate a peak reduction signal under power spectral density constraints. However, the accumulated MAI at the receiver can be significant, which leads to performance degradation. To address this problem, we consider both PAPR and MAI reduction using TR and propose ordering the subcarriers before allocating the peak reduction tones. In so doing, we are able to achieve PAPR reduction with causing only minimal MAI.

In this paper we investigate repeated root cyclic and negacyclic codes of length prm over Fpswith (m, p) = 1. In the case p odd, we give necessary and sufficient conditions on the existence of negacyclic self-dual codes. When m = 2m' with m' odd, we characterize the codes in terms of their generator polynomials. This provides simple conditions on the existence of self-dual negacyclic codes, and generalizes the results of Dinh [6]. We also answer an open problem concerning the number of self-dual cyclic codes given by Jia et al. [11].

In this paper, we consider the construction of lexi-codes by using a B-ordering over the ring Z4. This algorithm can be applied to any multiplicative property over Z4. It is shown that this construction produces many optimal codes over Z4and also good binary codes.

The Titchener T-complexity CTof a string has applications in, e.g., randomness testing, event detection and similarity comparison. Like the Lempel-Ziv production complexity, the upper bound of CTis demonstrably not a linear function of the string length. Knowledge of the bound for a given length is however required in order to convert CTinto a measure with linear upper bound such as Titchener's T-information. For this reason, the upper bound of CThas been investigated before by several authors, with various asymptotic solutions proposed. We present a new analytic closed-form asymptotic upper bound for CTbased on the Hurwitz-Lerch zeta function.

Node localization is employed in many wireless networks as it can be used to improve routing and enhance security. In this paper, we propose a new algorithm based on decision tree classification and K-means clustering which are well known techniques in data mining. Several performance measures are used to compare the K-means localization algorithm with those using linear least squares (LLS) and weighted linear least squares based on singular value decomposition (WLS-SVD). It is shown that the proposed algorithm performs better than the LLS and WLS-SVD algorithms even when the geometric anchor distribution about an unlocalized node is poor.

Differential pulse position amplitude modulation (DPPAM) is considered in an ultra wideband (UWB) communication system. DPPAM combines differential pulse position modulation (DPPM) and pulse amplitude modulation (PAM) to provide good performance with low computational complexity. The DPPAM UWB signal is derived from that of pulse position amplitude modulation (PPAM). The frame error rate (FER) of MN- ary DPPAM systems over additive white Gaussian noise (AWGN) and indoor fading channels is analyzed. The results show that the FER performance with 2N-ary DPPAM is better than that with 2N-ary DPPM and PPM, and MN-ary (M>;2) DPPAM provides a good compromise between FER and complexity.

Transmitted reference pulse cluster (TRPC) signalling was recently proposed for robust low-rate ultra-wideband (UWB) communications. TRPC provides significant performance gains over conventional transmitted reference (TR) UWB, and overcomes the critical long delay line problem associated with TR UWB. Thus, it is suited for practical implementation of a UWB communication system. The compact spacing of the pulses in TRPC leads to inter-pulse interference (IPI). In this study, the authors show that there exists an asymmetry between the two conditional distributions of the autocorrelation decision variable because of this IPI. This implies that the conventional zero decision threshold (ZDT) is not optimal for a TRPC autocorrelation receiver, and hence the authors derive the optimal maximum likelihood decision threshold. The authors propose two practical techniques for threshold determination. The first technique employs a training sequence whereas the other simply uses the decision variables of the received data signals to estimate the threshold. Simulation results show that in terms of the bit error rate (BER) performance, both the training-assisted adaptive decision threshold (TA-ADT) and the data-assisted ADT (DA-ADT) can achieve significant gains over the conventional ZDT-based detection for TRPC.

In this paper, we propose a speech-model based method using the linear predictive (LP) residual of the speech signal and the maximum-likelihood (ML) estimator proposed in “Blind estimation of reverberation time,” (R. Ratnam , J. Acoust. Soc. Amer., 2004) to blindly estimate the reverberation time (RT60). The input speech is passed through a low order linear predictive coding (LPC) filter to obtain the LP residual signal. It is proven that the unbiased autocorrelation function of this LP residual has the required properties to be used as an input to the ML estimator. It is shown that this method can successfully estimate the reverberation time with less data than existing blind methods. Experiments show that the proposed method can produce better estimates of RT60, even in highly reverberant rooms. This is because the entire input speech data is used in the estimation process. The proposed method is not sensitive to the type of input data (voiced, unvoiced), number of gaps, or window length. In addition, evaluation using white Gaussian noise and recorded babble noise shows that it can estimate RT60in the presence of (moderate) background noise.

This paper presents a method based on higher order statistics (HOS), namely the normalized third-order moment (skewness), for blind estimation of the inverse filter of the room impulse response (RIR). Skewness is used as a measure of asymmetry, and a comprehensive comparison with the commonly used metric (kurtosis) is presented. It is shown that a sufficiently long linear predictive (LP) residual of the speech signal has an asymmetric pdf with sufficient skewness to be used as a score function for the HOS-based approach. The proposed algorithm is optimized for the inverse filter estimation problem. This optimization includes an efficient initialization for high reverberation intensities, enabling the method to be employed in highly reverberant rooms. The direct-to-reverberation ratio (DRR) as well as the equalized impulse response clearly show that our method can estimate the inverse filter even in highly reverberant environments. In addition, performance results using recorded background noise and in time-varying environments illustrate that our approach is applicable in real world situations. The proposed method is shown to be superior to the method by Wu and Wang, particularly in terms of reducing the coloration effect. Experiments under different acoustic conditions confirm the effectiveness of the proposed method for time delay estimation (TDE). Finally, the proposed algorithm is used as the first-stage of monaural segregation, and it is shown to improve the performance under different conditions.

The Shannon code is a very simple suboptimal scheme for computing prefix-free codelengths for a given memoryless source. A recursive version of the well-known Shannon code (RSh) is investigated in this paper. It has a redundancy which never exceeds that of the Shannon code. Unlike the Huffman code, the RSh code and its variations can be directly applied to sources with an infinitely countable alphabet. The average redundancy, taken over the set of all n-tuple distributions, is considered as a criterion for code comparisons. For n>;40, the average redundancy is approximately 0.14 bits for the RSh code, compared to approximately 0.5 and 0.03 bits for the Shannon and Huffman codes, respectively. For large n, a constrained version of the RSh code, called CRSh, has an average redundancy of only 0.07 bits for unsorted symbol probabilities. If the symbol probabilities are sorted in descending order, then a very simple modification of the RSh algorithm results in a code which is near-optimal from the average redundancy point of view.

We study a single-source single-destination cooperative wireless communication system with multiple relays operating in a modified decode-and-forward mode. Nakagami-m fading with additive white Gaussian noise is assumed for all inter-node channels. The bursty data packet arrival is modeled by a Markov-modulated Poisson process (MMPP). A packet feedback model is proposed to characterize packet loss in the wireless channel. An approximation to the steady state distribution of the proposed generalized discrete time M/G/1-type queue at the source is obtained by state truncation. Packet level performance of four transmission modes adopting different modulation and coding schemes is analyzed under a variety of channel and traffic conditions using the truncated queueing analysis. The network power is used as a criterion to find boundary curves of adaptive modulation and coding (AMC) scheduling for the cooperative wireless system. Two source node transmission protocols, namely the transmit-every-clock-tick (TREC) and the LAZY protocols, are examined for both the AMC scheduling and channel utilization analysis.

Fitting noisy measurements to a circle is a classic statistical estimation problem. In this paper, we make two contributions to the study of this problem. First, we propose a novel formulation of the maximum likelihood (ML) estimator for identifying the center and radius of the circle from noisy measurements. This new estimator uses the unknown true values of the measurement points as the nuisance parameter to obtain an exact ML formulation. We then examine the Karush-Kuhn-Tucker (KKT) conditions for the optimum solution to the ML estimator. We show analytically that this new estimator is in fact equivalent to the well-known least squares (LS) form of the circle fitting problem. Second, from the insights gained in deriving the optimum solution, a computationally simple circle fitting algorithm based on greedy search is proposed. Performance results are given to illustrate the performance of the proposed algorithm.

With the limited wireless spectrum and the ever-increasing demand for wireless services, how to enlarge wireless network throughput is a pressing issue. To exploit the wireless spatial capacity, concurrent transmissions, if controlled appropriately, can lead to overall higher spectrum utilization and network throughput. The optimal scheduling and power control for concurrent transmissions in rate-adaptive wireless networks is a very challenging NP-hard problem. In this paper, we propose an efficient power allocation and scheduling algorithm for concurrent transmissions which can improve network throughput with fairness consideration. We first formulate the optimal power allocation and scheduling problem, and convert the original non-convex problem into a series of convex problems using a two-phase approximation technique. Then, we propose the power and channel allocation with fairness (PCAF) algorithm to solve the problem efficiently. Extensive simulation results show the remarkable improvement in terms of both network throughput and fairness, comparing to the previous scheduling algorithms.

A new class of high rate Fibonacci polynomial based error-correcting codes is introduced. For integers m; n; x ≥ 1, a square encoding matrix Q2mn(x) of order 2m and the corresponding decoding matrix Q2m-n(x) are provided. It is shown that a higher code rate is obtained than with the original Fibonacci polynomial based codes.

Node localization has many applications in wireless networks. For example, it can be used to improve routing and enhance security. Localization algorithms can be classified as range-free or range-based. Range-based algorithms use location metrics such as ToA, TDoA, RSS, and/or AoA to estimate the distance between nodes. Range free algorithms are based on proximity sensing. Range-based algorithms are more accurate but also more computationally complex. However, in applications such as target tracking, localization accuracy is important. In this paper, we propose a new range-based algorithm which is based on decision tree classification and the density based spatial clustering of applications with noise (DB SCAN) algorithm, which are well known in data mining. The Euclidean distance between intersection points is used as a distance metric, and the DBSCAN algorithm is applied to a subset of intersection points based on this metric. Different performance measures are used to compare our localization algorithm with linear least squares (LLS) and weighted linear least squares based on singular value decomposition (WLS SVD). The proposed algorithm is shown to perform better than the LLS and WLS-SVD algorithms even when the anchor geometric distribution about an unlocalized node is poor.

Femtocell networks have great potential for mobile applications. However, interference due to the co-existence of macrocells and femtocells is a serious problem. In addition, dense femtocells introduce severe inter-system interference. In this paper, a graph theory based dynamic sub-band allocation technique is presented to avoid downlink interference. We model the cells and their mutual interference as graph elements, nodes and weighted edges, respectively. To maintain a tolerable interference level, the total bandwidth is divided into a number of sub-bands, and these are assigned to the femtocells using a graph coloring algorithm. The division is optimized to minimize the bandwidth used to meet user traffic requirements, and minimize the femto-to-macro interference. In addition, sub-bands are assigned to the femtocells to avoid inter-femto interference. An iterative spectrum management algorithm is also introduced.

For q an odd prime and 1≤ m ≤ q, two specific binary qm × q2parity-check matrices denoted by HP(m, q) and HI(m, q) are considered. The corresponding binary codes, CP(m, q) and CI(m, q), respectively, are called proper and improper array codes with parameters m and q. Given a parity-check matrix H representing a binary code C, let s(H) denote the stopping distance of H and d(C) be the minimum Hamming distance of C. It is known that that s(HI(m, q)) = s(HP(m, q)) = d(CI(m, q)) = d(CP(m, q)) for m ≤ 3. In this paper, we show that these equalities do not hold for all values of m and q. In particular, although s(HP(4, 7)) = d(CP(4, 7)) = 8 we have s(HI(4, 7)) = 9 and d(CI(4, 7)) = 10. It is also shown that s(HP(5,1))dCP(5, 11)) = 10 while s(HI(5,11)) = 11 and d(CI(5, 11)) = 12. This suggests that in many cases the improper array codes would perform better than the proper array codes over the AWGN and binary erasure channels. Performance results are given which confirm this claim. The combinatorial structure of the eight-element stopping sets for H(m ≥ 4,q >; 5) is also determined.

This paper proposes a novel Dynamic Spectrum Management (DSM) scheme for Wideband Code Division Multiple Access (WCDMA) / Digital Video Broadcasting (DVB) heterogeneous systems. Capacity estimation algorithms for both WCDMA and DVB are developed which consider both the user distribution and characteristics of the hybrid services. Based on these algorithms, a new dynamic spectrum allocation scheme is presented which allows for optimum allocation of resources and maximum secondary spectrum usage. Coloring theory is used to significantly reduce DSM complexity while providing near-optimal performance. Numerical results are given which show that the proposed DSM scheme has better performance than Fixed Spectrum Management (FSM).

This paper proposes an authentication protocol between two parties. The protocol uses a shared secret key, representing a large variable-length code and a lookup table representing a permutation. The authenticating parties exchange randomly generated messages. Both challenge and response messages consist of multiple codewords from the code, and these messages are each truncated to a fixed number of symbols. The actual information to be recovered by the respondent are the codeword lengths used in the challenge message. These lengths are re-encoded to different lengths via the lookup table, and randomly chosen codewords with these re-encoded lengths are used in the response.

With the increasing convergence and cooperation among wireless networks, network awareness of user equipment (UE) has become very important. A practical solution for network information delivery for UE, Cognitive Pilot Channel (CPC), has recently been proposed. It can provide UE with the necessary network information for network selection by using the public signaling channel. In this paper, a cognitive multicast CPC scheme is proposed. It can greatly improve the on-demand CPC delivery efficiency, and also reduce the time delay of information delivery. The cognitive characteristics of the proposed multicast CPC scheme can easily be adapted to different heterogeneous network architectures.

Non-contiguous orthogonal frequency division multiplexing (NC-OFDM) is an effective transmission method for cognitive radio systems. It can provide high bandwidth utilization and robustness against time dispersive channels. However, NC-OFDM inherits the problems of high sidelobes and peak-to-average power ratio (PAPR) of the OFDM signal. If not accounted for, this introduces leakage into a primary user (PU) band and distorts the signal of the secondary user (SU). This paper proposes a joint sidelobe and PAPR reduction method that can simultaneously reduce the inherent OFDM out-of-band (OOB) radiation and the OOB radiation caused by a nonlinear power amplifier. In this method, selected mapping (SLM) sequences, which are generated from sequences with a low average sidelobe power, are used to improve the PAPR performance. The key idea is to generate multiple representations of the transmit signal and select a sequence with low OOB radiation and PAPR. We examine the performance of the proposed method taking into account the effects of non-ideal (i.e. non-linear) power amplification. The simulation results show that the proposed technique can significantly reduce OOB radiation while improving the PAPR and bit error rate (BER) for NC-OFDM signals.

GSM is the most popular standard for wireless cellular networks with 3 billion handsets in use worldwide. One of the critical components in the GSM architecture is the authentication protocol. The current protocol has several drawbacks such as one- way entity authentication, memory overhead in the VLR and bandwidth consumption between the HLR and VLR. Several solutions have been proposed to establish mutual entity authentication and to improve performance. However, they either cannot provide flaw-free bilateral entity authentication or impose more network overhead. In this paper, we propose a new approach to mutual entity authentication based on symmetric polynomials. In the proposed protocol, each MS is allocated a share of the symmetric polynomial during initial authentication. In subsequent authentication requests, the MS and VLR authenticate each other using the polynomial share. The proposed solution not only provides secure bilateral authentication, but also decreases the memory overhead in the VLR and the required connection bandwidth.

Sensor networks have been proposed for military and scientific applications such as border security and environment monitoring. They are usually deployed in unattended and hostile environments, so security is a major concern. A fundamental requirement in wireless network security is the ability to establish keys between pairs of sensors. In this paper, we propose a new location-based key management protocol in which polynomial-based and random key pre-distribution are both used for key establishment between sensor pairs. Key establishment between near sensors is provided by the polynomials, while key establishment between far sensors is accomplished by random key pre- distribution. Using these two approaches simultaneously reduces the overhead required for key establishment. Analysis is presented which shows that the proposed scheme has good performance compared with other approaches.

In this paper, a new technique for dynamic spectrum management of a Wideband Code Division Multiple Access (WCDMA)/Digital Video Broadcasting (DVB) heterogeneous system is introduced. The solution is obtained using coloring theory, and significantly reduces the complexity of spectrum management.

A greedy scheme called Greedy Codeword Assignment Scheme (GCAS) is proposed to assign D-ary codewords to the given code-lengths ¿1,¿2,...,¿n, so that they satisfy the fix-free property. This scheme guarantees that a D-ary fix-free code can be obtained whenever ¿i=1nD-¿¿ ¿(D), where ¿(D) is equal to 5/8 for D even and very close to 5/8 for D odd. This result can be regarded as an extension of Yekhanin's theorem on the existence of binary fix-free codes. In the special case D=2 , the greediness of GCAS enables us to prove that if mini ¿i= 2, the inequality ¿i=1n2-¿i¿ 21/32 implies the existence of a binary fix-free code with code-lengths ¿1,¿2,...,¿n.

Partial transmit sequence (PTS) is a proven technique to reduce the peak-to-average power ratio (PAPR) in orthogonal frequency division multiplexing (OFDM) systems. It achieves considerable PAPR reduction without distortion, but the high computational complexity of multiple Fourier transforms is a problem in practical systems. To address the complexity, signals at the middle stages of an ¿-point radix FFT using decimation in frequency (DIF) are employed for PTS subblocking. We formulate OFDM symbols based on these signals to exploit the periodic autocorrelation function (ACF) of the vectors in the PTS subblock partitioning. Error-correcting codes (ECCs) are employed in the subblocking for the PTS radix FFT. This new technique significantly decreases the computational complexity while providing comparable PAPR reduction to ordinary PTS (O-PTS), even with a small number of stages after PTS partitioning. Numerical results are presented which confirm the PAPR improvements.

A gradient algorithm has been used in the tone reservation technique for peak to average power ratio (PAPR) reduction. This provides a good approximation to the optimal solution with lower complexity. This technique requires that a kernel signal be computed and iteratively shifted in the time-domain to the peak locations to reduce the PAPR. The kernel signal should be updated dynamically for the best performance in time varying channels, which increases the computational complexity. In this paper, we propose a low complexity gradient algorithm for computing the peak reduction kernels. This approach is more efficient than the previous algorithm, and provides better PAPR performance.

Recently, a linear programming (LP) decoder has been introduced for binary linear codes. Although the performance of an LP decoder has a close relationship with the form of the parity check matrix (or equivalently with the Tanner graph) of the code, there is no clear approach to choosing a suitable form for LP decoding. In this paper, we focus on the class of cyclic codes, and show that the cyclic structure of the code can be used to expand the parity check matrix and obtain a low redundancy form which is suitable for LP decoding. Performance results are given which demonstrate the effectiveness of the proposed algorithm.

For a given memoryless information source, the Huffman code is the optimal prefix-free code in the sense of redundancy. Generally, the length of each codeword in the Huffman code is a function of all symbol probabilities p1, p2, ..., pn. In contrast, with the best known suboptimal code, i.e., the Shannon code, the length of the i-th codeword (i.e. [- log pi]) is a function of only pi. In this paper, two recursive versions of the Shannon code (RYY and RSh) are proposed which have redundancy which lies between that of the Huffman code and the Shannon code. In particular, the redundancy is not greater than that of the Shannon code and the i-th codeword length does not depend on pi+1, pi+2, ..., pn. In order to evaluate the overall performance of the proposed codes, their redundancy is considered as a random variable on the set of all sources with n symbols. An algorithm for generating random n-tuple distributions is derived and the expected value of the redundancy of the resulting codes is estimated. Recently, it was proven that the average redundancy of the Shannon code is around 0.5 bits. Simulation shows that for n > 20 the average redundancy of the proposed codes are about 0.1 and 0.06, while it is approximately 0.03 for the Huffman code.

Node localization is an essential component of many wireless networks. It can be used to improve routing and enhance the security of a network. Localization can be divided into rangefree and range-based algorithms. Range-based algorithms use measurements such as ToA, TDoA, RSS, and AoA to estimate the distance between two nodes. Range-free algorithms are based on proximity sensing between nodes. Although this is typically less accurate than range-based, it is cheaper and easier to implement. A new range-free scheme is proposed in this paper. Every target forms two sets of anchors. The first set contains one-hop anchors from the target. The second set contains two-hop and three-hop anchors away from the target. Each target uses the intersections between these anchors to estimate its position. This new approach is simple, but is more accurate than DRLS when the anchor ratio is low.

Selective mapping (SLM) has been proposed for peak to average power reduction (PAPR) in orthogonal frequency division multiplexing (OFDM) systems. With SLM, multiple sequences are generated by multiplying independent phase sequences with the original data and the sequence with the lowest PAPR is chosen for transmission. Hence, to determine the selected sequence at the receiver, side information must be sent along with the data. Previously, a maximum-likelihood (ML) decoder was proposed to eliminate this side information. However, this adds significant complexity to the receiver. In this paper, we first propose a partial SLM (P-SLM) technique using combinations of time-domain sequences. We then show that P-SLM can significantly reduce the ML decoder complexity. We examine the bit error rate (BER) performance of the simplified decoder over additive white Gaussian noise (AWGN) and fading channels. The results show that the proposed decoder has almost identical BER performance to that of the previous decoding algorithm while achieving very low computational complexity.

Despite extensive research on flat ad hoc networks, meeting satisfactory security levels in sensor networks presents many problems. The main reason is that these networks are typically designed based on hierarchal structures. In this paper, we propose an efficient scheme for authentication and key management in hierarchical ad hoc sensor networks using symmetric polynomials. In the proposed method, the processing requirements and traffic load are proportional to the processing capability and required bandwidth in each layer. An ad hoc node at the lowest level of the hierarchy uses symmetric cryptography. With more processing capabilities, entities in higher layers use public key cryptography. Performance results show the superiority of the proposed method in comparison with previous approaches developed for hierarchical ad hoc sensor networks.

Partial transmit sequence (PTS) is a proven technique for peak-to-average-power ratio (PAPR) reduction in orthogonal frequency division multiplexing (OFDM) systems. Determining the optimal phase factors requires an exhaustive search over all possible sets of phase factors. Hence, simplified optimal and suboptimal approaches have been proposed. Previous methods of formulating the phase factor optimization have considered variables over the entire OFDM symbol. With no degradation in performance, we show that fractional PTS sub-blocking can reduce the number of variables in the optimization problem. It can easily be employed with other techniques to significantly reduce complexity.

Selective mapping (SLM) has been employed to reduce the peak interference-to-carrier ratio (PICR) in OFDM systems. Here, we significantly reduce the computational complexity of SLM using subsets of intermediate signals within the inverse fast Fourier and fast Fourier transforms, while achieving performance close to that previously obtained.

To facilitate network performance analysis and simulations for IPTV traffic, a two-level Markovian traffic model is proposed in the paper. The model considers both spatial and temporal correlation in MPEG encoded video sequences, so it can mimic the highly variable data rate (VBR) behavior of IPTV sources. The model contains a Group of Pictures (GoP)- level Markov chain and a frame-level Markov chain, so it can capture both the inter-GoP and intra-GoP correlations. The proposed traffic model is simple to incorporate into network simulators, and can be used to obtain closed-form solutions of queue performance. Extensive simulations have been conducted to compare the network performance using the proposed model with the performance of a variety of real video traces. The results show that the accuracy of the proposed video source model is sufficient for the study of network performance. Therefore, it is an effective tool for performance evaluation of IPTV services via analysis and/or simulation.

One of major drawbacks of orthogonal frequency division multiplexing is sensitivity to frequency offsets caused by a mismatch between the transmitter and receiver oscillators. This offset destroys the orthogonality of the sub-carriers and introduces intercarrier interference (ICI), reducing the system performance. Previously, selective mapping was considered for reducing the peak interference-to-carrier ratio (PICR). However, this technique has a high computational complexity due to multiple inverse fast Fourier transform (IFFT) and fast Fourier transform (FFT) operations. In this paper, we exploit the IFFT/FFT structure to generate phase rotated SLM sequences and obtain a low PICR. This technique significantly reduces the computational complexity while providing a PICR performance close to that of the previous SLM technique.

Partial transmit sequence (PTS) is a proven technique to reduce the peak-to-average power ratio (PAPR) in orthogonal frequency division multiplexing (OFDM) systems. It achieves significant PAPR reduction without distortion, but the high computational complexity of multiple Fourier transforms is a problem in practical systems. In this paper, we consider this problem and use intermediate signals within a decimation in frequency (DIF) radix FFT to formulate the OFDM symbols. This enables us to exploit the periodic autocorrelation function (ACF) of the vectors in the PTS subblock partitioning. To improve the PAPR, we use repetition codes (RCs) in the subblocking for the PTS radix FFT. Our approach significantly decreases the computational complexity while providing comparable PAPR reduction to ordinary PTS (O-PTS). Numerical results are given which confirm our analysis.

Telecommunication service providers are racing to deliver IPTV/video on demand (VoD), voice, and data, the so called triple-play services. IPTV traffic, supported by the UDP protocol, has highly variable data rates and stringent quality of services (QoS) requirements in terms of delay and loss. Data and VoD flows are normally supported by TCP, which has its own congestion control loop to adjust the sending rate, so the traffic load is also highly dynamic. If IPTV and TCP traffic is simply multiplexed, their performance is difficult to predict and the competition between them will jeopardize their QoS. To efficiently utilize network resources and provide satisfactory QoS for both traffic types, we propose multiplexing IPTV and TCP traffic with the protection of a class based queuing (CBQ) scheme. We also develop an analytical framework to model the multiplexed IPTV traffic and TCP traffic with CBQ. The analytical results can be used as a guide to determine the admission region of IPTV and the CBQ parameters. Simulation results are presented which validate the analytical results and demonstrate the effectiveness of the proposed solution. By multiplexing IPTV and TCP traffic appropriately, network resources can be more efficiently utilized, the QoS of IPTV can be maintained, and TCP flows can obtain higher throughputs.

We consider the problem for which lengths a self-dual MDS code over Fqexists.We show that for q = 2m, there are self-dual MDS codes for all even lengths up to 2m. Furthermore, self-dual MDS codes of length q + 1 over Fqexist for all odd prime powers q. Additionally, we present some new self-dual MDS codes.

A new simple algorithm for OFDM symbol synchronization is proposed. The proposed algorithm does not assume that the first multipath component is the dominant one. Instead of using a threshold to detect the leading edge of an OFDM symbol, the proposed algorithm uses a metric which is calculated recursively. Two estimation methods are considered, one using the average of the metric results, and the other using the median. Performance results are given for two UWB channel models (CM1 and CM4), exponential, and Rayleigh fading channels which show that the proposed algorithm is superior to other algorithms, particularly at low signal to noise (SNR) ratios.

Orthogonal frequency division multiplexing (OFDM) modulation is widely used in communication systems to meet the demand for ever increasing data rates. Characteristics of the transmitted signal can be employed for blind channel identification. In this paper, we propose a blind polynomial channel estimation algorithm using noncircular second-order statistics of the received OFDM signal. A set of polynomial equations are then formulated based on the correlation of the received signal. The solution of these equations provides an estimate of the channel coefficients. Results are presented which show that the proposed algorithm provides performance comparable to the least minimum mean square error (LMMSE) solution with low computational complexity. The performance is near-optimal for large OFDM systems.

An IFFT-based selective mapping (SLM) technique is presented for reducing the peak-to-average power ratio (PAPR) of an orthogonal frequency division multiplexing (OFDM) signal. We use fractional inputs at intermediate stages within a decimation in time (DIT) IFFT to generate the SLM sequences. This significantly reduces the computational complexity while maintaining a PAPR performance close to that with original SLM. This new technique provides a better PAPR versus complexity tradeoff compared to previous techniques. Further, this approach provides similar complexity reduction for both low and high radix IFFT algorithms.

Orthogonal frequency division multiplexing (OFDM) modulation is widely used in communication systems to meet the demand for ever increasing data rates. In this paper, a low complexity joint semiblind detection algorithm for OFDM systems over time-varying channels is proposed based on the channel correlation and noise variance. The problem is relaxed to a continuous non-convex quadratic programming problem. Then an iterative method is utilized to deduce a sequence of reduced-size quadratic programming problems. These are solved by limiting the search in the 2-dimensional subspace. Furthermore, a low-bit descent search is employed to improve the system performance. Results are given which demonstrate that the proposed algorithm provides comparable performance with lower computational complexity than that of a sphere decoder.

The concatenation of marker and Reed–Solomon codes in order to correct insertion/deletion errors in differential pulse-position modulation (DPPM) over optical wireless communications is presented. The concatenated code decoding algorithms with hard-decision and soft-decision detection are presented. The performance of the hard-decision coded DPPM system is evaluated over both nondispersive and dispersive channels via analysis and simulation. It is shown that the coding gain provided by the concatenated code is approximately 4 dB when the code rate is about 0.7 and the channel is nondispersive. Over a dispersive channel, the coded system performs better than the uncoded system when the ratio of delay spread to bit duration is not high. A soft-decision detector is employed to combat intersymbol interference. The soft-decision decoding algorithm, which has low complexity and can be practically implemented, is described. The performance over nondispersive and dispersive channels is evaluated by analysis and simulation. It is shown that the soft-decision system requires approximately 2 dB less transmit power than the hard-decision system for additive white Gaussian noise and low-dispersive channels. Soft decoding also provides a performance improvement in high-dispersive channels.

In this correspondence, we construct new self-dual codes over ${\BBF}_4, {\BBF}_5, {\BBF}_7$ with larger minimum weights than the previously known self-dual codes, using construction methods based on circulant and negacirculant matrices. In particular, a self-dual $[36,18,13]$ code over ${\BBF}_7$ is constructed which has a larger minimum weight than the previously known $[36,18]$ codes.

We construct new MDS or near-MDS self-dual codes over large finite fields. In particular, we show that there exists a Euclidean self-dual MDS code of length n = q over GF(q) whenever q = 2m(m ges 2) using a Reed-Solomon (RS) code and its extension. It turns out that this multiple description source (MDS) self-dual code is an extended duadic code. We construct Euclidean self-dual near-MDS codes of length n = q-1 over GF(q) from RS codes when q = 1 (mod 4) and q les 113. We also construct many new MDS self-dual codes over GF(p) of length 16 for primes 29 les p les 113. Finally, we construct Euclidean/Hermitian self-dual MDS codes of lengths up to 14 over GF(q2) where q = 19, 23,25, 27, 29.

In this paper, we design rate-compatible punctured repeat-accumulate (RA) codes for the additive white Gaussian noise (AWGN) channel. The design involves optimizing the punctured profile of a mother code such that the resulting high rate codes give good performance. A three phase puncturing scheme is employed and the design employs EXIT charts. These codes are very simple in terms of complexity, and so are quite competitive to rate-compatible turbo or low density parity check (LDPC) codes. Codes with rates up to 10/11 are obtained from a single rate 1/3 regular RA code. Results show that these codes are good even at practical blocklengths such as 512 bits.

A heterogeneous wired and wireless network architecture is proposed for in-home IPTV distribution. To identify the bottleneck in the home network and estimate the network capacity, we develop an analytical framework to quantify the maximum number of IPTV connections that can be supported with guaranteed QoS in the wired and multi-hop wireless networks, respectively. We extend the fluid flow model analysis to capture both the burstiness of IPTV sources and the time-varying characteristics of multi-hop wireless channels. Extensive NS-2 simulations with H.264 HDTV sources over wired and multi- hop wireless paths are given, which validate the analysis. The analytical and simulation results provide important guidelines for the planning of future home networks and IPTV systems.

In a rapidly fading environment, Doppler spread caused by user mobility destroys the orthogonality among OFDM subcarriers, resulting in intercarrier interference. In this paper, a low complexity ICI reduction algorithm which can be applied to QAM signal constellations is proposed. A combinatorial optimization problem for ICI suppression is formulated and then relaxed to a quadratic programming problem. A successive method is then utilized to deduce a sequence of reduced-size QP problems, which is solved by limiting the search in the 2- dimensional subspace spanned by its steepest-descent and Newton directions to reduce the computational complexity. Furthermore, a low-bit descent search is employed to enhance the system performance. The proposed algorithm is shown to provide excellent performance with low computational complexity.

One of the most important and challenging problems in coding theory is to construct codes with the best possible parameters. Quasi-cyclic (QC) and the larger class of quasi- twisted (QT) codes have been proven to contain many good codes (with best-known parameters). In this paper, we review some open problems concerning these codes, introduce generalizations of QT codes, and suggest some constructions involving QT codes. We also present some new and good quaternary codes.

It is well known that any two non-identical codewords from a fixed-length code must belong to different non- periodic equivalence classes under cyclic shift for the code to have a bounded synchronization delay (BSD). Subsets of fixed- length codewords from T-codes, a family of variable-length codes (VLCs) known for strong statistical self-synchronization and a well-explained synchronization model, also follow this rule, albeit with some periodic exceptions. This paper shows how the T-code construction algorithm may be used for the simple construction of maximal block codes with bounded synchronization delay. These self-synchronizing codes are subsets of T-codes and thus resynchronization under decoding with a T-code decoder is both automatic and detectable.

A low complexity partial transmit sequence (PTS) technique for reducing the peak-to-average power ratio (PAPR) of an orthogonal frequency division multiplexing (OFDM) signal is presented. Signals at the middle stages of an Appoint radix IFFT are considered for PTS subblocking. We use decimation in frequency (DIF) and propose a low complexity technique, called decomposition PTS (D-PTS) subblocking, where subblocks are assigned through different stages of the transform. This new technique reduces the multiplicative complexity, while providing PAPR reduction similar to other techniques such as ordinary PTS (O-PTS). Moreover, it has lower additive complexity.

We consider an IFFT-based technique for reducing the peak-to-average power ratio (PAPR) of an orthogonal frequency division multiplexing (OFDM) signal. The intermediate signals in a radix decimation-in-time (DIT) algorithm is used for subblock partitioning in the partial transmit sequence (PTS) technique. We present a new subblocking technique which significantly reduces the computational complexity while providing PAPR reduction comparable to that with ordinary PTS.

A radix FFT-based technique for reducing the peak-to-average power ratio (PAPR) of an orthogonal frequency division multiplexing (OFDM) signal is presented. The authors use intermediate signals in radix decimation in time (DIT) or frequency (DIF) FFT algorithm for subblock partitioning in the partial transmit sequence (PTS) technique. It is shown that DIF achieves low multiplicative complexity in generating PTSs compared to that with DIT and the same PAPR reduction. In addition, high radix FFT algorithms provide better PAPR reduction performance per stage with less multiplicative complexity compared with low radix algorithms.

In this paper, we present rate-compatible systematic repeat-accumulate (RA) codes for the additive white Gaussian noise (AWGN) channel. The systematic form is used because of the higher degree parity checks we require in the code. We show that very high code rates can be attained with good performance through our puncturing schemes. Although RA codes are very simple in terms of complexity compared to other codes such as turbo codes or low density parity check (LDPC) codes, the performance of these codes is quite competitive. Codes with rates up to 9/10 are obtained from a single rate 1/3 systematic regular RA code. Performance results show that our puncturing provides superior performance at high code rates compared to just puncturing parity bits.

This paper presents two main results relating to elliptic curve cryptography. First, a double point compression scheme is proposed which allows a compact representation of elliptic curve points without the computational cost associated with ordinary single point compression. A triple point compression scheme is also proposed which can result in more savings in memory and/or bandwidth. Second, a new approach to speeding up random point multiplication is given for the case where the base point is variable but available in a certificate. In this approach, some redundant information (a few multiples of the base point) is added to the certificate. It is shown that a significant speed up can be obtained by optimizing the Moller's algorithm for the case where only a portion of the lookup table is available. It is also shown how to use redundant information to compute random point multiplication using parallel processors. The proposed point compression schemes can be employed to reduce the required bandwidth when single point compression is computationally expensive

The problem of selecting a code for finite monotone sources with N symbols is considered. The selection criterion is based on minimizing the average redundancy (called Minave criterion) instead of its maximum (i.e., Minimax criterion). The average probability distribution PNmacr, whose associated Huffman code has the minimum average redundancy, is derived. The entropy of the average distribution (i.e., H(PNmacr)) and the average entropy of the monotone distributions (i.e., H(PNmacr)) are studied. It is shown that both logN-H(PNmacr) and logN-H(PNmacr) are asymptotically equal to a constant (sime0.61). Therefore, there is only a negligible penalty (at most 1.61 bits/symbol) in using a simple fixed-length code with respect to the optimal code. An efficient near-optimal encoding technique is also proposed. The consequences of the two approaches, i.e., Minave and Minimax, are compared in terms of their associated distributions and associated codes. In order to evaluate the average performance of the Minimax code, we prove that the informational divergence of the average distribution and Minimax distribution asymptotically grows as -2.275+loglogN

An orthogonal frequency division multiplexing (OFDM) system has the problem of the peak-to-average power ratio (PAPR) due to the inter symbol interference between the many sub-carriers. In general, in order to obtain optimal PAPR reduction using the partial transmitted sequence (PTS), the total search for the number of sub-blocks and the rotation factors must be accomplished. As the number of sub-blocks and rotation factors increases, PAPR reduction improves. The number of calculation increases as the number of sub-blocks increases, such that complexity increases exponentially and the process delay occurs simultaneously. In order to reduce the complexity in the system, in this paper a new method using GA is proposed to search for a rotation factor that reduces simultaneously both the PAPR and the amount of calculation, such that the complexity of calculation and the process time are reduced at the same time. Comparison is performed between the proposed method and the various techniques developed previously. The superiority of proposed method is presented by demonstrating the reduction of complexity and calculation time while a similar PAR reduction is obtained.

Previously, self-dual codes ternary have been constructed from conference matrices. In this paper, we present codes constructed from negacirculant conference matrices. A necessary condition for these codes to be self-dual is given, and examples are given for lengths up to 108. The equivalence with the Pless symmetry codes is established

The minimax distribution PFNdefined by the probabilities pi,FN= 1/lambdaN(i-1)(i-1)/iiplays an important role in the context of encoding finite monotone sources with unknown probabilities. The Shannon code for this distribution is a suboptimal minimax code. The average performance of this suboptimal code is considered in this paper. In order to evaluate the average performance of the minimax code, we compare it with the minave code which minimizes the average redundancy over all monotone sources with N symbols. To achieve this, we study the informational divergence of the minave and minimax distributions and prove that it asymptotically grows as -2.275+log log N. The log log N degradation in average performance should not be regarded as a drawback of the minimax code because the minimum average codeword length over the class of monotone sources grows as log N, which asymptotically dominates log log N

In this paper, we introduce a new multi-mode Beyond 3rdGeneration (B3G) access system concept based on heterogeneous physical layer modes or configurations, with a common (or anchor) layer 2 and 3 protocols stack. Such a system can support a wide variety of physical layer access techniques, e.g. Code Division Multiple Access (CDMA), Orthogonal Frequency Division Multiple Access (OFDMA), Multi-Carrier CDMA (MC-CDMA) etc, with different configurations which are optimized for different deployment scenarios and performance targets, while anchored by a common layer 2 and 3 protocols. Seamless inter-system handoff is enabled by the common layer 2/3 protocols, while dynamic radio resource/load/spectrum management can be performed across the different physical layer modes to achieve optimum spectrum efficiency and Quality of Service (QoS) support. The overall system concept of the multimode access system is first presented, followed by a detailed account of the anchor layer 2/3 protocol design.

In this paper, we study the problem of optimal pulse shaping for pulse position modulation (PPM) ultra-wideband (UWB) systems with a recently proposed sparsity-driven signal detection method. This signal detection method offers superior performance over traditional matched filter based detection through the representation of additive white Gaussian noise (AWGN) and the UWB pulses via atoms from a Hadamard Walsh matrix and a pulse constellation dictionary, respectively. Recognizing the possibility to further improve the performance with sparsity-driven signal detection through shaping the UWB pulses to be dissimilar to AWGN, we formulate an optimal pulse shaping problem considering the federal communication commission (FCC) emission mask. We also develop a relaxation method to approximate the objective function, and solve the relaxed problem with classical nonlinear programming. Design examples are given to show the resulting pulse shape, its dissimilarity to the channel noise, and its compliance with the mandatory FCC emission mask.

Ultra-Wideband (UWB) transmission is a promising technology for high speed wireless networks such as wireless personal area networks (WPANs). The detection of UWB signals is a key issue for system performance and capacity. In this paper, we propose a novel signal detection technique for multiple access UWB systems. Our approach is based on an overcomplete dictionary signal representation as well as the concept of successive interference cancellation for multi-user detection. We introduce the algorithm for multiple access UWB signal detection, and present guidelines to generate overcomplete dictionaries for the detection of signals in noise. The proposed detection algorithm for multiple access UWB systems can eliminate noise as well as multi-user interference (MUI). Signal detection is achieved by solving an l1norm minimization problem, which can be calculated using low complexity linear programming algorithms.

In this paper, we propose a soft-decision decoding algorithm for differential pulse-position modulation (DPPM) over optical wireless communications. We describe the soft decoding algorithm which has low complexity and can be practically implemented. The performance over nondispersive and dispersive channels is evaluated. It is shown that soft decoding requires approximately 2 dB less transmit power than hard decoding in low dispersive AWGN channels. Soft decoding also provides a performance improvement in high dispersive channels.

This paper presents the concatenation of marker and Reed-Solomon codes in order to correct insertion/deletion errors in differential pulse-position modulation (DPPM) over wireless infrared communications. The concatenated code decoding algorithm is presented. We evaluate the performance of the coded DPPM system over both nondispersive and dispersive channels via analysis and simulation. It is shown that the coding gain provided by the concatenated code is approximately 4 dB when the code rate is about 0.7 and the channel is nondispersive. Over a dispersive channel, the coded system performs better than the uncoded system when the ratio of delay spread to bit duration is low. However, the power penalty increases faster when this ratio is high

Ultra-wideband (UWB) transmission is a promising technology for future high speed wireless networks. In this paper, we propose a novel signal detection approach with noise suppression for UWB systems. This approach is based on an overcomplete dictionary signal representation and a sparsity-driven optimization algorithm. We introduce general approaches for UWB signal detection, and present guidelines to generate overcomplete dictionaries for signals in noise. UWB signal detection is achieved by solving an l1norm minimization problem, which can be solved using low complexity linear programming algorithms. Examples are given to demonstrate the performance of the proposed approach

In this letter, system capacity and error probability of orthogonal space-time block coding (STBC) is considered for PAM/PSK/QAM modulation in correlated Nakagami fading channels. The approach is based on an equivalent scalar AWGN (additive white Gaussian noise) channel with a channel gain proportional to the Frobenius norm of the matrix channel for the STBC. Closed form capacity and error probability expressions are derived for Nakagami fading channels. Numerical results are given to illustrate the theory.

Recently, pulse position amplitude modulation (PPAM) has been proposed for ultra-wideband (UWB) communication systems. PPAM combines pulse position modulation (PPM) and pulse amplitude modulation (PAM) to provide better performance and higher system capacity with low computational complexity. A RAKE receiver can make use of the rich multipath of UWB systems to improve system performance and capacity. In this paper we present the performance and capacity of a time hopping (TH) UWB system with PPAM modulation and RAKE reception over a practical multipath fading channel. Error probability and a union bound on performance, single user capacity and a capacity/distance bound based on FCC Part 15 rules, are derived. This analysis is extended to the multiple access case. Numerical results are given to illustrate the analysis.

A new metric, the multivariate merit factor (MMF) of a Boolean function, is presented, and various infinite recursive quadratic sequence constructions are given for which both univariate and multivariate merit factors can be computed exactly. In some cases these constructions lead to merit factors with non-vanishing asymptotes. A formula for the average value of 1/MMF is derived and a characterisation of the MMF in terms of cryptographic differentials is discussed.

In this paper we consider the performance of a RAKE receiver in a code division multiple access (CDMA) system employing maximal-ratio combining. A simple select and combine algorithm is introduced in which the RAKE receiver combines only the synchronization path and those resolvable paths with a signal-to-noise ratio (SNR) larger than a given threshold. We analyze its performance based on moment generating functions of the SNR of the combined signal. We study the cases with equal and unequal average SNR for different diversity paths. In particular, closed-form expressions for the average combined SNR and symbol error probability with the proposed RAKE receiver over block fading channels are derived. We also study the outage probability of the system and the number of channels estimated. Numerical results are given to illustrate the analytical results.

In this paper, the error probability of orthogonal space-time block coding (STBC) is considered in correlated flat fading channels. The approach is based on an equivalent scalar AWGN (additive white Gaussian noise) channel. Closed form error probability expressions are derived for PSK/PAM/QAM on correlated fading channels.

Quality-of-service (QoS) is a desirable feature for mobile ad hoc networks (MANETs) due to the growth of multimedia applications. However, the mobile nature and dynamic topology of MANETs make it difficult to provide QoS assurance in such networks. In this paper we propose a QoS routing protocol based on AODV (QS-AODV), which creates routes according to application QoS requirements. A local repair mechanism is used to improve the packet delivery ratio. It is shown that QS-AODV provides performance comparable to AODV under light traffic conditions. In heavy traffic, QS-AODV provides higher packet delivery ratios and lower routing overheads, at a cost of slightly longer end-to-end delays. The effects of mobility on performance is also presented.

In this paper, the capacity of orthogonal space-time block coding (STBC) is considered for PAM/PSK/QAM modulation in correlated flat fading channels. The approach is based on an equivalent scalar AWGN (additive white Gaussian noise) channel with a channel gain proportional to the Frobenius norm of the matrix channel for STBC.

The feasibility of converting text into speech using an inexpensive computer with minimal memory is of great interest. Speech synthesizers have been developed for many popular languages (e.g., English, Chinese, Spanish, French, etc.), but designing a speech synthesizer for a language is largely dependant on the language structure. In this article, we develop a Persian synthesizer that includes an innovative text analyzer module. In the synthesizer, the text is segmented into words and after preprocessing, a neural network is passed over each word. In addition to preprocessing, a new model (SEHMM) is used as a postprocessor to compensate for errors generated by the neural network. The performance of the proposed model is verified and the intelligibility of the synthetic speech is assessed via listening tests.

This paper introduces a new radix-2 representation with the same average weight as the width-w nonadjacent form (w-NAF). In both w-NAF and the proposed representations, each nonzero digit is an odd integer with absolute value less than M. However, for w-NAF, M is of the form 2/sup w-1/, while, for the proposed representation, it can be any positive integer. Therefore, using the proposed integer representation, we can use the available memory efficiently, which is attractive for devices with limited memory. Another advantage of the proposed representation over-w-NAF is that it can be obtained by scanning the bits from left-to-right. This property is also useful for memory-constrained devices because it can reduce both the time and space complexity of fast point multiplication techniques.

The iterative decoding structure and component maximum a posteriori decoders used for decoding binary concatenated codes can be extended to the nonbinary domain. This paper considers turbo codes over nonbinary rings, specifically ternary, quaternary, penternary, hexernary, and octernary codes. The best rate-1/2 component codes are determined using a practical search algorithm. The performance of the resulting rate-1/3 turbo codes on an additive white Gaussian noise channel using q-ary phase-shift keying modulation is given.

In this letter, we propose a new modulation scheme called pulse position amplitude modulation (PPAM) for ultra-wideband (UWB) communication systems. PPAM combines pulse position modulation and pulse amplitude modulation to provide good system performance and low computational complexity. The channel capacity of PPAM is determined for a time-hopping multiple-access UWB communication system. The error probability and performance bounds are derived for a multiuser environment.

In this paper, we propose a new modulation scheme called biorthogonal pulse position modulation (BPPM) for ultra-wideband (UWB) communication systems. A set of N=2/sup k+1/ BPPM signals are constructed from 2/sup k/ orthogonal PPM signals by including the antipodal version of the orthogonal PPM signals. The channel capacity of BPPM is determined for a time-hopping multiple access UWB communication system. The error probability and performance bounds are derived for a multiuser environment. It is shown that N-ary BPPM has better performance than N-ary PPM with the same throughput and half the computational complexity.

We propose a novel differential amplitude pulse-position modulation (DAPPM) for indoor optical wireless communications. We present the symbol structure and the properties of DAPPM. The power spectral density of DAPPM is also derived and compared to other modulation schemes. DAPPM yields advantages over PPM and DPPM in terms of bandwidth efficiency, capacity and peak-to-average power ratio (PAPR). The power efficiency is also examined. Over a non-dispersive channel, DAPPM gives better bandwidth and/or power efficiency depending on the number of amplitude levels (A) and the maximum length (L) of a symbol. We also show that, over a dispersive channel, DAPPM has better bandwidth efficiency but requires more power than PPM and DPPM for the same value of L.

The Shannon capacity of orthogonal space-time block codes (STBC) is considered for Rayleigh, Ricean and Nakagami-m uncorrelated fading channels and correlated Rayleigh fading channels. In this paper, STBC capacity expressions are derived for several fading channels. The Shannon capacity of a multiple antenna wireless system over a fading channel with continuous-valued inputs and continuous-valued outputs can be obtained.

In this paper, we develop a simple closed-form series expression for the error probability of maximum ratio combining (MRC) multichannel reception of M-ary coherent systems over flat Ricean fading channels. This series expansion provides a simple and numerically efficient mean of calculating the error probability with controlled accuracy. M-ary PAM, PSK and QAM systems are investigated.

In this paper, we propose a new modulation scheme called pulse amplitude position modulation (PPAM) for ultra-wideband (UWB) communication systems. PPAM combines pulse position modulation (PPM) and pulse amplitude modulation (PAM) to provide good system performance and low computational complexity. A set of MN-ary, M=2/sup k/, N=2/sup N/. PPAM signals are constructed from N-ary orthogonal PPM signals by including M-ary PAM signals in each dimension. It is shown that MN-ary PPAM has better performance than MN-ary PAM and less complexity than MN-ary PPM for MN>2. The channel capacity of PPAM is determined for a time-hopping multiple access UWB communication system. The error probability and performance bounds are derived for a multiuser environment. In particular, it is shown that for M=2. 2N-ary PPAM signals have better performance than 2N-ary PPM with the same throughput and half the computational complexity.

It is well known that the problem of finding stabilizer quantum-error-correcting codes (QECC) is transformed into the problem of finding additive self-orthogonal codes over the Galois field GF(4) under a trace inner product. Our purpose is to classify the extremal additive circulant self-dual codes of lengths up to 15, and construct good codes for lengths 16/spl les/n/spl les/27. We also classify the extremal additive 4-circulant self-dual codes of lengths 4,6,8,12,14, and 16 and most codes of length 10, and construct good codes of even lengths up to 22. Furthermore, we classify the extremal additive bordered 4-circulant self-dual codes of lengths 3,5,7,9,11,13,15, and 17, and construct good codes for lengths 19,21,23, and 25. We give the current status of known extremal (or optimal) additive self-dual codes of lengths 12 to 27.

We present a method for maximum likelihood decoding of the (48,24,12) quadratic residue code. This method is based on projecting the code onto a subcode with an acyclic Tanner graph, and representing the set of coset leaders by a trellis diagram. This results in a two level coset decoding which can be considered a systematic generalization of the Wagner rule. We show that unlike the (24,12,8) Golay code, the (48,24,12) code does not have a Pless-construction which has been an open question in the literature. It is determined that the highest minimum distance of a (48,24) binary code having a Pless (1986) construction is 10, and up to equivalence there are three such codes.

We classify all optimal linear [n,n/2,d] codes over F/sub 4/ up to length 18. In particular, we show that there is a unique optimal [12,6,6] code and three optimal [16,8,7] codes, up to equivalence.

This article investigates the asymptotic performance of single parity-check (SPC) product codes (PCs) from a decoding point of view. Specifically, the probability of bit error is bounded before and after the decoding of each dimension, similar to the analysis of "iterated codes" by Elias (1954). It is shown that the asymptotic probability of bit error can be driven to zero as the number of dimensions, and hence the block length, increases at signal-to-noise ratios (SNRs) within 2 dB of capacity over the additive white Gaussian noise (AWGN) channel.

In this paper, the average probability distribution P~/sup N/~ of equiprobable finite monotone sources, where N is the number of symbols, is derived and some properties of this distribution are investigated. The associated Huffman code has the minimum redundancy in the average sense. An efficient suboptimal encoding technique is proposed. We show that log N -H(P~/sup N/~) is a constant asymptotically, and consequently there is a negligible penalty (asymptotically) in using a fixed length code. The length of the shortest codeword should be modified such that lim/sub N/spl rarr//spl infin// l/sub N//l/sub 1/=2. This limit cannot be a constant for minimax and universal codes such as the /spl omega/ and /spl delta/ codes. These properties are also shown to hold when the number of symbols is unknown but bounded.

This paper considers the performance of iteratively decoded single parity check (SPC) multidimensional product codes in an additive white Gaussian noise channel. Asymptotic performance bounds are compared to simulation results. A new code structure based on SPC product codes is introduced. This structure involves interleaving between the encoding of each dimension in the product code. An analysis of the weight distribution is used to explain the good performance results for these randomly interleaved SPC product codes.

Turbo-codes are decoded using iterative decoding algorithms with somewhat elusive convergence properties. In this paper, the modes of convergence observed in an extensive series of turbo-code simulations are categorized by examining the bit convergence behavior for each frame. Based on these results, a new method for decoder termination based on average log-likelihood ratios is presented and compared with other methods. As an application, a selective repeat automatic repeat request (ARQ) system is considered.

Previously, self-dual codes have been constructed from Hadamard matrices. In this correspondence, codes constructed from weighing matrices, and in particular conference matrices are presented. A necessary condition for these codes to be self-dual is given, and examples are given for lengths up to 40. Codes constructed from all weighing matrices of order n/spl les/13 are also considered.

Previously, Type II codes over F/sub 4/ have been introduced as Euclidean self-dual codes with the property that all Lee weights are divisible by four. In this paper, a number of properties of Type II codes are presented. We construct several extremal Type II codes and a number of extremal Type I codes. It is also shown that there are seven Type II codes of length 12, up to permutation equivalence.

Let [n, k, d]/sub q/-codes be linear codes of length n, dimension k, and minimum Hamming distance d over GF(q). In this article, 31 new quaternary (q=4) codes are constructed which improve the respective lower bounds on minimum distance in Brouwer's tables.

Optimal double circulant self-dual codes over F/sub 4/ have been found for each length n/spl les/40. For lengths n/spl les/14, 20, 22, 24, 28, and 30, these codes are optimal self-dual codes. For length 26, the code attains the highest known minimum weight. For n/spl ges/32, the codes presented provide the highest known minimum weights. The [36,18,12] self-dual code improves the lower bound on the highest minimum weight for a [36,18] linear code.

This paper presents the asymptotic performance of Hamming and extended Hamming product codes for the binary symmetric channel and the additive white Gaussian noise (AWGN) channel. Simulation results for very low complexity high dimensional single parity check product codes on the AWGN channel are also given.

Let [n,k,d;q]-codes be linear codes of length n, dimension k, and minimum Hamming distance d over GF(q). In this correspondence, 18 codes are constructed which improve the known lower bounds on minimum distance.

In this correspondence, we give a characterization of certain quasi-cyclic self-complementary codes with parameters [120,9,56] and [136,9,64], some quasi-cyclic self-complementary codes are also constructed with parameters [496,11,240] and [528,11,256]. These codes are optimal in the sense that they meet the Grey-Rankin bound, new quasi-symmetric SDP (symmetric difference property) designs are constructed from these codes.

Previously, Type IV self dual codes over the ring F/sub 2/+uF/sub 2/ of order 4 have been introduced. In this paper, we construct five optimal Type IV self-dual codes which are the first examples of such codes for that length. We also give several new optimal type IV self-dual codes, and Type IV self-dual codes which improve the upper bounds for minimum weights.

In 1976, Moore investigated a class of self-dual double-circulant codes based on primes of the form q=3 mod 8. This paper shows that the corresponding length 168 code contains a weight 28 codeword and thus is not extremal. This answers a long-standing question regarding this code. The next five codes in this class are also shown not to be extremal.

Given a quasi-cyclic code with minimum Hamming distance d, a set of convolutional codes is derived with free distance equal to d. It is shown that an increase in the rate of these codes results in a decrease in the memory length. The connection between these codes is illustrated with several well-known quasi-cyclic codes. The free distance of some partial unit memory convolutional codes can be determined using the results in this correspondence.

Recently there has been tremendous interest in self-dual codes over finite rings, specifically the rings Z/sub 2k/. In this paper, we investigate double circulant self-dual codes over Z/sub 2k/ for small k and short lengths. In particular, we give a classification of the extremal double circulant codes. Using invariant theory, a basis for the space of invariants to which the Hamming weight enumerators belong is given for self-dual codes over Z/sub 6/, Z/sub 8/, Z/sub 10/, and Z/sub 12/.

In this correspondence, the relationship between iterative decoding and techniques for minimizing cross-entropy is explained. It is shown that minimum cross-entropy (MCE) decoding is an optimal lossless decoding algorithm but its complexity limits its practical implementation. Use of a maximum a posteriori (MAP) symbol estimation algorithm instead of the true MCE algorithm provides practical algorithms that are identical to those proposed in the literature. In particular, turbo decoding is shown to be equivalent to an optimal algorithm for iteratively minimizing cross-entropy under an implicit independence assumption.

This paper considers the combination of multiple copies of a packet to improve the performance of a slotted direct-sequence spread-spectrum multiple-access (DS/SSMA) ALOHA packet radio system with coherent binary phase-shift keying (BPSK) modulation. Both slotted DS/SSMA ALOHA with and without forward error correction (FEC) are considered. For the case with FEC, maximum-likelihood decoding with code combining is used. Code combining allows for the combination of multiple copies of the same packet (which are typically discarded), to obtain a lower code rate for that specific packet, and therefore an improved probability of successful decoding. In both cases, combining multiple copies of the same packet results in a throughput which is an increasing function over a broad range of offered load, so that the system is more reliable from the point of view of stability. In addition, combining provides a higher throughput and a smaller time delay for packet transmission. This is illustrated by means of analytical and simulation results.

Reed-Solomon (RS) error-correcting codes are often proposed for communication systems requiring burst and/or erasure correction capabilities. In most cases, the modulation symbol size is fixed a priori. Therefore, the effective application of RS coding implies proper matching of the code symbol size with the modulation symbol size. Previous results have assumed that the RS codeword symbol size is an integer multiple of the modulation symbol size or have been based on simple approximations. In this paper, the exact symbol error probability with K-bit M-ary modulation symbols and q-bit RS codeword symbols is presented.

Let [n,k,d;q]-codes be linear codes of length n, dimension k and minimum Hamming distance d over GF(q). The following quasi-cyclic codes are constructed in this paper: [44,11,20;3], [55,11:26:3], [66,11,32;3], [48,12,21;3], [60,12,28;3], [56,13,24;3], [65,13,29;3], [56,14,23;3], [60,15,23;3], [64,16,25;3], [36,9,19;4], [90,9,55;4], [99,9,61;4], [30,10,14;4], [50,10,27;4], [55,10,30;4], [33,11,15;4], [44,11,22;4], [55,11,29;4], [36,12,16;4], [48,12,23;4], [60,12,31;4]. All of these codes have established or exceed the respective lower bounds on the minimum distance given by Brouwer.

In this correspondence, we investigate binary extremal self-dual codes. Numerous extremal self-dual codes and interesting self-dual codes with minimum weight d=14 and 16 are constructed. In particular, the first extremal Type I [86,43,16] code and new extremal self-dual codes with weight enumerators which were not previously known to exist for lengths 40,50,52 and 54 are constructed. We also determine the possible weight enumerators for extremal Type I codes of lengths 66-100.

The largest quasi-cyclic subcode of the Reed-Muller code R(r,m), invariant under the shift T/sup 2m-l/ is determined. This code, denoted QCR(r,m,l), is presented through its module decomposition into cyclic submodules. The smallest regular trellis diagram (SRTD) is defined for block codes. This trellis and its construction algorithm are given for the class of cyclic-form codes. Using the cyclic-form structure of QCR(r,m,l), the 2/sup l/-section SRTD of this code is determined. The eight-section SRTD is given for the Reed-Muller codes R(r,m). The quasi-cyclic subcodes of R(r,m) with regular 2/sup l/-section minimal trellis diagrams are presented.

New ternary codes of dimension 6 are presented which improve the bounds on optimal linear codes. These codes belong to the class of quasi-twisted (QT) codes, and have been constructed using a greedy algorithm. This work extends previous results on QT codes of dimension 6. In particular, several new two-weight QT codes are presented. Numerous new optimal codes which meet the Griesmer bound are given, as well as others which establish lower bounds on the maximum minimum distance.

Eighteen new codes are presented which improve the bounds on maximum minimum distance for binary linear codes. They are rate (m-r)/pm, r/spl ges/1, degenerate quasi-cyclic (QC) codes. Based on the known upper bounds, six of these new codes are optimal. In addition, five two-weight QC codes of dimension 8 are given.

New codes of dimension 7 are presented which give improved bounds on the maximum possible minimum distance of ternary linear codes. These codes belong to the class of quasi-cyclic codes, and have been constructed using a stochastic optimization algorithm, tabu search. Thirty-two codes are given which improve or establish the current bounds for ternary codes. In addition, a table of upper and lower bounds for d/sub 3/(n, 7) is presented for n/spl les/240.

A frequency-hopped spread-spectrum (FH/SS) packet radio network is considered. Instead of discarding the received packets which were unsuccessfully decoded, the receiver keeps and combines the different received copies of the same packet to improve the throughput and reduce the delay necessary to transmit a packet successfully. It is shown that the throughput is substantially increased over a similar system where the uncorrectable packets are discarded. This system is equivalent to a fast frequency-hopped system where the diversity level is adapted according to the channel conditions. Reed-Solomon (RS) codes are used for error correction and detection.

In this correspondence, new optimal quaternary linear codes of dimension 5 are presented which extend previously known results. These codes belong to the class of quasi-twisted (QT) codes, and have been obtained using a greedy local search algorithm. Other codes are also given which provide a lower bound on the maximum possible minimum distance. The generator polynomials for these codes are tabulated, and the minimum distances of known QT codes are given.

Conway and Sloane (1990) have listed the possible weight enumerators for extremal self-dual codes up to length 72. In this correspondence, we construct extremal singly-even self-dual [60,30,12] codes whose weight enumerator does not appear in this list. In addition, we present the possible weight enumerators for extremal self-dual codes of length 60.

The class of quasi-twisted (QT) codes is a generalization of the class of quasi-cyclic codes, similar to the way constacyclic codes are a generalization of cyclic codes. In this paper, rate 1/p QT codes over GF(3) are presented which have been constructed using integer linear programming and heuristic combinatorial optimization. Many of these attain the maximum possible minimum distance for any linear code with the given parameters, and several improve the maximum known minimum distances. Two of these new codes, namely (90, 6, 57) and (120, 6, 78), are optimal and so prove that d/sub 3/(90, 6)=57 and d/sub 3/(120, 6)=78.>

Investigates the performance of a hybrid DS/SFH spread spectrum multiple access communication system using BPSK modulation over a mobile radio channel in the presence of interference and noise. A decision feedback equalizer (DFE) which provides inherent diversity in a multipath environment is utilized at the receiver. The DFE is also known to significantly reduce cochannel interference and intersymbol interference. Simulation results are presented with minimum mean square error (MMSE) and bit error rate as performance measures.>

This paper considers the application of rate-adaptive coding (RAC) to a spread spectrum multiple access (SSMA) communication system. Specifically, RAC using a variable rate Reed-Solomon (RS) code with a single decoder is applied to frequency-hopped SSMA. We show that this combination can accommodate a larger number of users compared to that with conventional fixed-rate coding. This increase is a result of a reduction in the channel interference from other users. The penalty for this improvement in most cases is a slight increase in the delay (composed of propagation and decoding delay). The throughput and the undetected error probability for a M-ary symmetric channel are analyzed, and performance results are presented.>

The performance of M-ary differential phase shift keying (MDPSK) in both tone jamming and Gaussian noise is presented. Previous work has been extended by considering larger signal sets and skewed differential phases with unequal decision regions. The error performance when a frequency offset exists between the jamming tone and MDPSK carrier is also considered.>

A class of rate 2/P quasi-cyclic codes can be characterized in terms of m/spl times/m circulant matrices. In this correspondence, two new codes with parameters (80, 10, 35) and (95, 10, 42) are presented which improve the known lower bound on the maximum possible minimum distance. The former code can be extended with an even parity check bit to an (81, 10, 36) code that establishes that d/sub 2/(81, 10)=36.>

A method for obtaining the probability density function (PDF) and the cumulative density function (CDF) of sum-independent random variables is presented. The method is capable of determining the PDF and CDF of this sum for an input consisting of any combination of a signal tone, white Gaussian noise, and multiple interfering tones. It is based upon circularly symmetric function theory, Fourier-Bessel series, and Fourier series. To illustrate this method, applications are presented for a fast frequency-hopped noncoherent frequency-shift-keyed communications system. From the PDF and CDF of the received signal, performance values such as the error probability for demodulation, the probability of detection and false alarm for coarse-time synchronization and the mean and variance of timing-error estimates for fine-time synchronization are obtained.>

A double error correcting systematic (16,8) quasi-cycle (QC) code that can detect all triple-adjacent errors within each 8-b byte is presented. This code is useful in computer memory applications where adjacent errors are more likely than random errors. As an alternative, a systematic (24,16) QC code is given which corrects single errors, and detects all double errors in addition to all triple-adjacent and quadruple-adjacent errors. A third code, a systematic (32,16) QC code, is presented which corrects all triple errors and detects all quadruple errors. Decoding can be achieved either by syndrome generation and table lookup or by majority logic.>

Twelve new quasi-cyclic (QC) codes are presented which improve the bounds on the best known binary linear codes. They are rate (m-r)/pm, r>1, codes constructed by deleting r rows from the circulant matrices of rate m/pm QC codes, based on the factors of x/sup m/-1. Heuristic combinatorial optimization was used to find these codes.>

Self-reciprocal polynomials (SRPs) over GF(q), where q is a prime power, q=p/sup k/, are investigated. The maximum possible component for these polynomials is found for q odd. The construction of Fermat maximum exponent self-reciprocal polynomials (MRPs) over GF(2) is extended to GF(2/sup k/) with the aid of generalized Fermat numbers. These polynomials leads to a bound on the maximum possible exponent of SRPs over GF(2/sup k/), and a simplified algorithm for finding these MRPs. Self-reciprocal polynomials have applications in cryptography, error-correction coding, and the synthesis of linear feedback shift registers. They are advantageous when available memory or hardware is restricted or when data can be read in either direction. Some results on quasi-self-reciprocal polynomials are also presented.>

The class of quasi-cyclic (QC) codes has been proven to contain many good codes. To date the known results are largely codes of the form 1/p and (p-1)/p constructed from circulant matrices. A generalization of these rate 1/p codes to rate (m-1)/pm codes based on the theory of 1-generator QC codes is presented. The results of a search for good codes based on heuristic combinatorial optimization are nine codes which improve the known lower bounds on the minimum distance of binary linear codes.>

The class of quasi-cyclic (QC) codes has been proven to contain many good codes. To date the known QC codes are primarily rate 1/p and (p-1)/p binary codes constructed from circulant matrices. These results are extended to QC codes over GF(3) and GF(4). Codes are constructed using integer linear programming and heuristic combinatorial optimization. Many of these attain the maximum possible minimum distance for any linear code with the same parameters, and several improve the maximum known distances. The link between power residue (PR) codes and QC codes is exploited as a means of constructing new QC codes and to initialize the search algorithm. Previously unknown minimum distances for nonbinary quadratic residue codes are given. The minimum distances for the PR codes and the maximum known distances for the QC codes are tabulated.>

Tables of 1/p and rate (p-1 )/p binary quasi-cyclic (QC) codes that extend previously published results are presented. Many of these codes attain bounds given by T. Verhoeff (1987), who composed a table of bounds on the maximum possible minimum distance of binary linear codes. Many of the codes presented meet or improve these bounds. A best code is considered to be one which has the largest possible minimum distance for the given code dimensions. n and k, and class of error correcting codes. A good code has the maximum known minimum distance for the class of codes. An optimal code is one that achieves the maximum possible minimum distance for a linear code with the same dimensions. Binary power residue codes are found and used to construct QC codes.>

The probability distribution of the received DPSK (differential phase-shift keying) signal under tone jamming is studied. The results should facilitate the analysis of the SFH/DPSK (slow frequency-hopped DPSK) system. The results given are more general than those previously published, in several aspects. First, the differential phase of the transmitted DPSK signal can assume any value. Second, probability distributions are derived instead of a set of probabilities calculated over certain symmetrical regions. This allows analyzing performance for arbitrary selected decision regions as well as determining optimum decision regions for demodulating the DPSK signal. Third, the joint probability distribution of both the magnitude and differential phase of the jammed DPSK signal is given. This can be used in the analysis where both tone jamming and Gaussian noise are considered. These results are used to analyze the error probability performance of a general uncoded SFH/DPSK signal under worst-case tone jamming and Gaussian noise.>

The performance of fast-frequency-hopped M-ary frequency-shift keying with a fixed hop rate is evaluated, utilizing the Chernoff union bound method. The performance criterion used is a throughput measure i.e., an information rate sustained by a system for a given bit error rate, normalized by the hop rate. Both uncoded and coded systems are considered. It is shown using the cutoff rate argument that coding can provide a few dB gain in throughput. This is confirmed by the performance evaluation of various convolutional and block codes. Both partial-band noise jamming and multitone jamming with one tone per M-ary band are considered. Jamming parameters are assumed to be the worst case against the coding channel. Determination of the optimum M is also addressed.>