高可靠系统的电源电压监控和定序

对于多数电子系统,用上电复位(POR)电路监控系统电压能保证正确的上电初始化.用POR监测降低电压跌路也使程序运行问题(可使存储器不可靠或导致系统不能正确运行)最少.为了改善高端系统的可靠性,系统电源必须有正确的时序,以防止其微控制器,微处理器,DSP或ASIC闭锁,闭锁问题可能导致系统损坏或长期可靠性差.在大多数情况下,一个或多个微处理器监控IC可实现这些定序和监控功能.     

Approximate sensitivity analysis for acyclic Markov reliability models

Acyclic Markov chains are frequently used for reliability analysis of nonmaintained mission-critical computer-based systems. Since traditional sensitivity (or importance) analysis using Markov chains can be computationally expensive, an approximate approach is presented which is easy to compute and which performs quite well in test cases. This approach is presented in terms of a Markov chain which is used for solving a dynamic fault-tree, but the approach applies to any acyclic Markov reliability model.     

Modeling VoIP Call Holding Times for Telecommunications

Voice over IP is one of the most popular applications in broadband access networks. It is anticipated that the characteristics of call holding times (CHTs) for VoIP calls will be quite different from traditional phone calls. This article analyzes the CHTs for mobile VoIP calls based on measured data collected from commercial operation. Previous approaches directly used the Kolmogorov-Smirnov (K-S) test to derive the CHT distributions, which may cause inaccuracy. In this article we propose a new approach to derive the CHT distributions for mobile VoIP calls and other call types. Specifically, our approach uses hazard rate to select an appropriate distribution, and then utilizes the K-S test to validate our selection. We show that the mobile VoIP CHT distribution can be accurately approximated by a mix of two log-normal distributions. Based on the derived distributions, we compare the mobile VoIP CHTs with those for non-VoIP calls and fixed-network VoIP calls. Our study indicates that the characteristics for mobile VoIP calls are quite different from those of the non-VoIP mobile phone calls and are more close to those of fixed-network phone calls.     

Low-Complexity Digital Tone Detector for Telecommunications Switching Systems

An improved structure for an envelope detection multifrequency receiver based on digital nonrecursive quadrature filters is proposed. The receiver utilizes a property of digital nonrecursive quadrature filters to reduce the required computational complexity by a nominal factor of 2 with a negligible hardware increase. The application to the tone detector implementation for two CCITT standard signaling systems is considered, showing the low computational and structural complexity required by the proposed technique. The required multiplication rate is reduced by more than twice with respect to other conventional approaches.     

Hidden Markov Modeling of Error Patterns and Soft Outputs for Simulation of Wideband CDMA Transmission Systems

In this paper, a Hidden Markov Modeling (HMM) technique for a fast and accurate simulation of bit errors and soft outputs in wireless communication systems is presented. HMMs with continuous probability distributions are considered. Soft outputs and bit errors are combined to error patterns. We focus on binary phase–shift keying (BPSK) modulation for direct–sequence spread spectrum (code–division multiple access, CDMA) transmission as proposed e.g. for the third generation wireless communication system UMTS (uplink for the frequency division duplex mode (FDD)). Comparisons of simulated bit error rates for HMM models and Rake receivers are shown for AWGN, flat fading, and vehicular channel conditions. In order to assess the ability of the HMM to describe the dynamical behaviour of the channel a comparison for transmission with interleaving and convolutional coding is presented. Furthermore calculated autocorrelation functions of the error patterns and error gap distributions corresponding to the Rake receiver and to the HMM, respectively, are presented. Our investigations show a strong dependence of the required HMM order on E_b/N₀ and the channel conditions. The degree of accordance of the HMM outputs and the training data is examined based on calculated statistical scoring indicators.     

The Challenge of the VLSI Technique to Telecommunications Systems

The rapid evolution from the transistor to the LSI circuit revolutionized telecommunications systems. In particular, the low-cost IC's triggered the transition from wired- to stored-program control and from analog to digital techniques by introducing standard IC's as micro-processors, memories, etc. The availability of analog-digital interface circuits, for example codecs, filters, SLIC's, and vocoders, has become very important. In addition to the technical revolution, the economical aspects must be considered. LSI circuits enter more and more into the vicious circle of greater complexity-fewer applications and thus higher costs. An escape from this circle is offered by programmable circuits such as microprocessors or new circuits with a regular architecture. Nevertheless, custom circuits will always be used in order to protect the system know-how. The blend of standard and custom circuits will be important for the success of a systems house. Since vertical integration activities will be unlikely, different concepts of cooperation between IC manufacturers and system house become more and more important.     

Modeling iCAR via Multi-Dimensional Markov Chains

iCAR is a new wireless system architecture based on the integration of cellular and modern ad hoc relaying technologies. It addresses the congestion problem due to limited channel access in a cellular system and provides interoperability for heterogeneous networks. The iCAR system can efficiently balance traffic loads and share channel resource between cells by using ad hoc relaying stations (ARS) to relay traffic from one cell to another dynamically. Analyzing the performance of iCAR is nontrivial as the classic Erlang-B formula no longer applies when relaying is used. In this paper, we build multi-dimensional Markov chains to analyze the performance of the iCAR system in terms of the call blocking probability. In particular, we develop an approximate model as well as an accurate model. While it can be time-consuming and tedious to obtain the solutions of the accurate model, the approximate model yields analytical results that are close to the simulation results we obtained previously. Our results show that with a limited number of ARSs, the call blocking probability in a congested cell as well as the overall system can be reduced.     

Universal Filtering Via Hidden Markov Modeling

The problem of discrete universal filtering, in which the components of a discrete signal emitted by an unknown source and corrupted by a known discrete memoryless channel (DMC) are to be causally estimated, is considered. A family of filters are derived, and are shown to be universally asymptotically optimal in the sense of achieving the optimum filtering performance when the clean signal is stationary, ergodic, and satisfies an additional mild positivity condition. Our schemes are comprised of approximating the noisy signal using a hidden Markov process (HMP) via maximum-likelihood (ML) estimation, followed by the use of the forward recursions for HMP state estimation. It is shown that as the data length increases, and as the number of states in the HMP approximation increases, our family of filters attains the performance of the optimal distribution-dependent filter. An extension to the case of channels with memory is also established.     

动态系统辨识的一种近似算法

利用函数与其导数的方块脉冲展开式系数之间的近似公式，给出了一种动态系统辨识的方法，并证明了将该方法应用于单输入单输出系统时的精度及收敛性。     

Open Architecture Standard for NASA's Software-Defined Space Telecommunications Radio Systems

NASA is developing an architecture standard for software-defined radios used in space- and ground-based platforms to enable commonality among radio developments to enhance capability and services while reducing mission and programmatic risk. Transceivers (or transponders) with functionality primarily defined in software (e.g., firmware) have the ability to change their functional behavior through software alone. This radio architecture standard offers value by employing common waveform software interfaces, method of instantiation, operation, and testing among different compliant hardware and software products. These common interfaces within the architecture abstract application software from the underlying hardware to enable technology insertion independently at either the software or hardware layer. This paper presents the initial Space Telecommunications Radio System architecture for NASA missions to provide the desired software abstraction and flexibility while minimizing the resources necessary to support the architecture.     

Approximate Output Processes in Hidden-User Packet Radio Systems

The processes consisting of the packet interdeparture times for contention-type packet broadcasting systems in a hidden-user, singlehop environment are studied under the heavy-traffic assumption. The channel access protocols considered include pure ALOHA and unslotted nonpersistent carrier-sense-multiple-access (CSMA). The theory of superposition of independent renewal processes is applied to approximate the distribution of the duration of each unsuccessful transmission period in channel state. Our analysis results for the channel throughput and the coefficient of variation for the packet interdeparture time in symmetric configurations are shown to be in good agreement with simulation results over a wide range of offered channel traffic.     

Reduced Complexity Dynamic Systems Using Approximate Control Moments

This paper deals with reduction of computational complexities in dynamic systems. This paper develops a novel method of reducing complexities with use of control moments of the system. Though the proposed method is validated through channel estimation in this paper, the same can be equally applied to any other dynamic systems. Encouraging results given in this paper prove that the computational complexities can be reduced up to 10⁴ with a marginal affordable loss of performance.     

Approximate Modeling of Optical Buffers for Variable Length Packets

This paper addresses the problem of dimensioning buffers realized by means of fiber delay lines in optical routers able to switch packets that have variable length and are sent asynchronously on the optical links. The optical buffer is analyzed focusing on the different behavior of a delay buffer and an electronic memory. The role of the time unit of the fiber delay lines is discussed, showing that it is a crucial parameter to determine the queuing performance. The paper presents two approximate analytical models that can be used both for analysis and engineering of the optical buffer and in particular to dimension the buffer time unit in an way that is optimal with respect to packet loss probability. The first model is based on an infinite queuing approximation. It is not very accurate and is valid for a limited set of values of the traffic load, but is extremely simple. The second model is based on a finite queuing approximation. It is more complex but more accurate and is valid for any value of traffic load. The accuracy of the models is compared with simulation and their range of applicability purposes is discussed.     

基于马尔可夫切换系统的固定时滞平滑算法

提出了一种基于马尔可夫切换系统的固定时滞平滑算法,在交互式多模型结构框架内引入状态增广矩阵,将目标状态的当前值和过去值相结合,使状态预测与平滑算法同步进行,算法还应用当前观测数据估算时滞模型概率,方法简便易行。通过仿真对比跟踪性能可以得出：固定时滞平滑算法的性能要优于标准IMM EKF滤波算法,并随着时滞长度的增加,性能趋向更优。     

Statistical Estimation of Fast Varying Channels for Cellular Mobile Telecommunications Systems

The time variant environment in fast channels severely impacts on the performances of classical equalization techniques. A method is proposed here to circumvent this problem, particularly in the severe channel with fast behaviour. To improve the Bit Error Rate (BER) at the receiver, a new approach is proposed for the channel impulse response (CIR) estimation in the Enhanced Data rates for GSM Evolution (EDGE)cellular system. The computation of the CIR h(t) (finite impulse response filter) is performed by using the Least Squares algorithm. Then, a statistical estimation of the CIR is achieved to update the coefficients of the equalizer filter. Computer simulation results show that statistical estimation (CIR re-estimation) provides significant improvement with 250 km/h vehiclespeed (RA250): the bit error rate (BER) is 20% lowered for data bits far from the middle of the burst.     

Modeling of Block-Based DSP Systems

Modeling semantics based on dataflow graphs are used widely in design tools for digital signal processing (DSP). This paper develops efficient techniques for representing and manipulating block-based operations in dataflow-based DSP design tools. In this context, a block refers to a finite-length sequence of data items, such as a sequence of speech samples, an image, or a group of video frames, as part of an enclosing data stream. We develop in this paper a meta-modeling technique called blocked dataflow (BLDF) for augmenting DSP design tools with more effective blocked data support in an efficient and general manner. We compare BLDF against alternative modeling approaches through a detailed case study of an MPEG 2 video encoder system.Dong-Ik Ko received the B.S. and M.S. degree from Korea University, Korea, in 1993 and in 1995, all in electrical and computer engineering. He had been with LG Electronics, Korea as an embedded system develpoer from 1996 to 2001.He is now a Ph.D. student of Electrical and Compuetr Engineering, University of Maryland College Park, MD. His resarch interests include hardware/software co-design, design methodology for embedded systems, and performance and memory issues in system-level synthesis.Shuvra S. Bhattacharyya is an associate professor in the Department of Electrical and Computer Engineering, and the Institute for Advanced Computer Studies (UMIACS) at the University of Maryland, College Park. He is also an affiliate associate professor in the Department of Computer Science. Dr. Bhattacharyya is coauthor or coeditor of three books and the author or coauthor of more than 70 refereed technical articles. His research interests include VLSI signal processing, embedded software, and hardware/software co-design. He received the B.S. degree from the University of Wisconsin at Madison, and the Ph.D. degree from the University of California at Berkeley. Dr. Bhattacharyya has held industrial positions as a Researcher at the Hitachi America (San Jose, California) Semiconductor Research Laboratory, and as a Compiler Developer at Kuck & Associates (Champaign, Illinois).     

基于Swarm平台的装备体系建模研究

对Swarm平台进行了简介,它是基于复杂适应系统的工具集,而装备体系属于复杂系统并可以用复杂系统的方法来解决问题,构建了多Agent/CA模型框架并对其关键技术进行了论述,最后介绍了仿真流程。在此创新地将CAS理论引入装备体系对抗建模领域,并基于Swarm平台建立了装备体系建模框架。     

基于着色Petri网的电信业务特性集成模型

“业务特性”的快速集成是电信系统发展的趋势,本文提出了基于着色Ｐｅｔｒｉ网（ＣＰ网）的业务特性集成模型。在本方法中,基本的电话系统以及每个业务特性以一个ＣＰ网来模拟,当把一个业务特性集成到系统中时,相应于把代表它的ＣＰ网嵌入系统的网中。本模型既形象化地模拟业务特性的集成,同时可以用数学方法来分析业务特性的行为,可应用于检测和解决业务特性的交互作用。     

Modeling burst channels using partitioned Fritchman's Markov models

Discrete models based on functions of Markov chains (also referred to as hidden Markov models or finite-state channel (FSC) models) have been used to characterize the error process in communication channels with memory. One important property of these models is that the probability of any observed sequence can be expressed as a linear combination of the probability of a finite set of sequences of finite length, the so-called basis sequences. In this paper, we express the parameters of a class of FSC models as a simple function of the probability of the basis sequences. Based on this approach, we propose a new method for the parameterization of the Fritchman (1967) channel with single-error state as well as the interesting cases of Fritchman channels with more than one error state and the Gilbert-Elliott channel ((GEC) nonrenewal models). To illustrate the method, FSC models for the nonfrequency-selective Rician fading channel are presented. The number of states and the probability of state transitions are estimated for a given set of fading parameters     

Fading Channel Modeling via Variable-Length Markov Chain Technique

Channel characterization and modeling are essential to the wireless communication system design. A model that optimally represents a fading channel with a variable-length Markov chain (VLMC) is proposed in this paper. A VLMC offers a general class of Markov chains whose structure has a variable order and a parsimonious number of transition probabilities. The proposed model consists of two main components: 1) the optimal fading partition under the constraint of a transmission policy and 2) the derivation of the best VLMC representation with respect to the Kullback-Leibler (K-L) distance of fading samples. The fading partition is used to discretize a continuous fading channel gain. The optimal discretization criterion is developed based on the cost function of fading channel statistics and the transmission policy used in the system. Once a continuous fading channel gain is discretized, a VLMC is then used to model the channel. To obtain the optimal VLMC representation, we use the K-L distance of the discretized fading samples as the optimization criterion. The K-L distance of the discretized fading samples is used to determine the appropriate transition probabilities characterizing the optimal VLMC. Last, we show simulation results that demonstrate the accuracy and the effectiveness of the proposed fading channel representation in modeling the Rayleigh fading as well as the lognormal fading.