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The article presents several research papers relating to algorithms. The paper "Greatest Common Divisor of n Integers and Multipliers" presents an algorithm which calculates the greatest common divisor, IGCD, of n integers. Details of the method and comparisons to other algorithms are also given. The algorithm is a new version of the Euclidean algorithm for n integers. The n-1 calculations of the greatest common divisor of two integers is accomplished by means of a modified version of the Blankinship algorithm. The paper "Exponential Integral Ei (x)" presents the results of one phase of research carried out at the Jet Propulsion Laboratory, California Institute of Technology, under Contract NAS7-100, sponsored by the National Aeronautics and Space Administration. It presents an algorithm which was compiled and executed without any modification on a UNIVAC 1108 computer. An unfortunate precedent has been set in several recent algorithms of using an illegal FORTRAN construction.

Consider a multipriority batch computer system which users from several different classes may join, with toll, service, and waiting charges. Such a system is formulated here as a semi- Markov decision process, In which the aim of arriving users Is to minimize their expected loss. The optimal joining policy of arriving users who may join the system at some of its queues is a control limit policy, with a single control number for any possible queue and the user's class; a newly arriving user will join a queue that is not filled up to the control number corresponding to this queue and the user's class. In this paper control numbers, as well as lower and upper bounds for the control numbers and the capacities of the system's queues, are derived. [ABSTRACT FROM AUTHOR]

Purpose – The purpose of this paper is to develop a new recognition system for identifying advanced radar emitter signals (RES). Design/methodology/approach – Initially, the framework of the new recognition system is outlined. Then, feature extraction using resemblance coefficient and wavelet packet decomposition, and feature selection based on quantum-inspired genetic algorithm, and a classifier combining K-means clustering, support vector machines and Mahalanobis distance are applied to actualize the recognition system. Finally, experiments are carried out on RES. Findings – A valid recognition system with its framework and implementation is presented to solve the difficult problem of advanced RES recognition. Research limitations/implications – Initial investigation is made on modern RES recognition. Further work may be done on decreasing the error rates and enhancing recognition efficiency. Real signals instead of simulated signals can be applied. Practical implications – The system developed here can be applied to electronic reconnaissance systems such as electronic support measures, electronic intelligence and radar warning receiver. Originality/value – The paper presents a novel recognition system and its implementation for modern RES. Extensive experiments conducted on 155 RES with eight intra-pulse modulations show the feasibility and validity of the introduced system. [ABSTRACT FROM AUTHOR]

In this paper a novel online calibration technique for high-speed DACs is presented. The approach consists of two elements. The first element is the use of a redundant signed digit (RSD) scheme for the selection of the current sources. This enables a fully digital correction. The second element consists of an adaptive estimation of the correction terms by an LMS algorithm. For this purpose the DAC output signal is low-pass filtered and digitized by an accurate but low-speed calibration ADC. By replicating this analog signal path in the digital domain, the errors can be calculated and used to update the correction weights. The dynamic behavior and the accuracy of the adaptive calibration loop are analyzed both theoretically and through computer simulations, and it is shown that this way a greatly improved accuracy can be obtained. [ABSTRACT FROM AUTHOR]

In this paper, a new feedforward symbol timing recovery technique using timing estimation followed by interpolation is presented for digital receivers with two samples/symbol or higher sampling rate. A few timing estimation algorithms are proposed to estimate the timing offset accurately. The basic algorithm uses only the in-phase (I) or quadrature (Q) signal for timing detection, which applies to a BPSK communication system. It is shown that the basic algorithm, when applied in quadrature modulation systems where both I and Q signals are available, can be modified slightly to yield an improved estimation precision. The mean and variance of the resulting timing estimate are analyzed and simulated, supporting a satisfactory estimation performance. It is also shown that by applying a postprocessing scheme, such as the Kalman filter, the variance can be further reduced, resulting in a smoothed timing estimate. Some of the issues concerning the implementation of the proposed technique are also addressed. [ABSTRACT FROM AUTHOR]

In this paper, we present a hierarchical method that decomposes a discrete-event system (DES) into a high-level subsystem which communicates with n ⩾ 1 parallel low-level subsystems, through separate interfaces which restrict the interaction of the subsystems. It is a generalization of the serial case (n = 1) described in Part I of this paper, where we define an interface and a set of interface consistency properties that can be used to verify if a DES is nonblocking and controllable. Each clause of the definition can be verified using a single subsystem; thus the complete system model never needs to be stored in memory, offering potentially significant savings in computational resources. We provide algorithms for verifying these new properties, and briefly discuss the computational complexity of the method. Finally, we present an application to a large manufacturing example with an estimated worst-case closed-loop state-space size of 2.9 × 1021. [ABSTRACT FROM AUTHOR]

Purpose – The purpose of this paper is to present the development and comparison of selected time-domain and Laplace-domain methods for the simulation of waves propagating along multiconductor transmission lines (MTLs), both uniform and nonuniform, and sensitivities with respect to distributed and lumped parameters of MTL systems. Design/methodology/approach – A methodology is based on discrete, semidiscrete and continuous MTL models formulated and solved both in the time and Laplace domains, latter combined with a numerical inverse Laplace transform (NILT). Findings – The most accurate method is that based on the MTL Laplace-domain continuous model, processed via the MTL chain matrix and connected with an NILT. This method concurrently shows minimal RAM requirements, and in case of uniform MTLs, it runs fastest. For nonuniform MTLs, however, the implicit Wendroff formula is fastest, as long as the RAM is available. Research limitations/implications – The research is limited to linear MTLs only and the methods suppose terminating circuits based on their generalized Thévenin equivalents. They can be, however, generalized for more complex systems via more sophisticated boundary conditions treatment. The time-domain methods have further potential to be generalized towards nonlinear MTLs. Practical implications – The methods considered can contribute to solving signal integrity issues in high-speed electronic systems, the Matlab routines developed can serve in education process as well. Originality/value – The implicit Wendroff formula has been adapted to enable simulation of voltage and/or current distributions and their sensitivities along the nonuniform MTLs' wires. Besides, semidiscrete and continuous nonuniform MTL models have been elaborated to enable sensitivities determination, both in the time and Laplace domains, latter connected with the NILT technique based on fast Fourier transform/inverse fast Fourier transform and quotient-difference algorithms. [ABSTRACT FROM AUTHOR]

Abstract: Quality of service (QoS) is a measure of a communication network performance that reflects the transmission quality and service availability as perceived by the users. In the context of telecommunications, voice communication quality is the most visible and important aspects to QoS, and the ability to monitor and design for this quality should be a top priority. Voice quality refers to the clearness of a speaker''s voice as perceived by a listener. Its measurement offers a means of adding the human end-user''s perspective to traditional ways of performing network management evaluation of voice telephony services. Traditionally, measurement of users'' perception of voice quality has been performed by expensive and time-consuming subjective listening tests. Over the last decade, numerous attempts have been made to supplement subjective tests with objective measurements based on algorithms that can be computerised and automated. This paper examines some of the technicalities associated with voice quality measurement, and presents a review of current subjective and objective voice quality measurement methods and standards as applied to telecommunication systems and devices, with particular focus on recent and internationally standardised methods. [Copyright &y& Elsevier]

The performance of feature-aided tracking (FAT) relies largely on the quality of features extracted from signature sensors. In our previous work, we introduced a new FAT algorithm for tracking with ground moving target indicator (GMTI) and high resolutional range (HRR) measurements, where features were extracted from an HRR sensor using the technique of mixture density estimation. Although satisfactory results were achieved, an additional improvement is expected if a target rigidity constraint is incorporated. In this paper, we exploit the rigidity property of targets to alleviate the inherent local convergence problem of the expectation-maximization algorithm used in mixture density estimation. Simulation results show a performance improvement over the previous FAT algorithm. [ABSTRACT FROM AUTHOR]

Since complex network theory was first put forward, the search issue for networks has drawn increasing attention from multidisciplinary researchers, and has played an important role in network study. Many practical applications require search algorithms such as searching for the shortest relationship link in social networks, seeking web sites on the Internet, and finding specified files in data sets. The key issue for these tasks is how to achieve a fast search. However, traditional random-walk-based methods cannot solve this problem effectively. In order to fast travel, a novel search algorithm which employs a highest degree likelihood approach with k hunters looking for the target simultaneously is presented for different types of complex networks. A state transition matrix is applied to explain this proposed method. We compare the proposed algorithm with the methods of forerunner in the simulation, and the results show that our algorithm performs more effectively. Finally, some applications and future challenges are discussed. [ABSTRACT FROM AUTHOR]

The modified discrete cosine transform (MDCT) and inverse MDCT (IMDCT) are two of the most computationally intensive operations in MPEG audio coding standards. A new mixed- radix algorithm for efficiently computing the MDCT/IMDCT is presented. The proposed mixed-radix MDCT algorithm is com- posed of two recursive algorithms. The first algorithm, called the radix-2 decimation-in-frequency algorithm, is obtained by decomposing an N-point MDCT into two MDCTs with the length N/2. The second algorithm, called the radix-3 decimation-in-time algorithm, is obtained by decomposing an N-point MDCT into three MDCTs with the length N/3. Since the proposed MDCT algorithm is also expressed in the form of a simple sparse matrix factorization, the corresponding IMDCT algorithm can be easily derived by simply transposing the matrix factorization. Comparison of the proposed algorithm with some existing ones shows that our proposed algorithm is more suitable for parallel implementation and particularly suitable for the layer III of MPEG-1 and MPEG-2 audio encoding and decoding. Moreover, the proposed algorithm can be easily extended to the multidimensional case by using the vector-radix method. [ABSTRACT FROM AUTHOR]

We present an improved analog floating-gate pFET synapse that implements a supervised leaning algorithm similar to the least mean square (LMS) Learning rule. Weight decay plays a key role in several learning rules; this floating-gate synapse exhibits this behavior. We examine implications of the weight decay appearing in the correlation learning rule realized in the floating-gate synapse and provide experimental data characterizing the synapse and its performance in one-input and two-input LMS networks. Analog floating-gate synapses will enable larger-scale, on-chip learning networks than previously possible. [ABSTRACT FROM AUTHOR]