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IEEE Global Telecommunications Conference GLOBECOM '91: Countdown to the New Millennium. Conference Record最新文献

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Synchronization technique for optical OOK receivers 光学OOK接收机的同步技术
V. Jain, A.B. Asgill, P. Bobrek
The authors present a purely digital timing recovery scheme for optical OOK receivers, which offers the advantages of low hardware complexity and low jitter. In this technique, the computation of the timing function is based on symbol-rate samples of the received signal. A gain scheduling strategy is used for the timing updates; a short training set of symbols accompanied by large gain for the timing updates is followed by data symbols and low gain for the timing updates. This allows the use of large correction step sizes during training, leading to fast convergence. It is particularly suitable for an all digital optical system.<>
提出了一种用于光OOK接收机的纯数字时序恢复方案,该方案具有硬件复杂度低、抖动小等优点。在该技术中,时序函数的计算是基于接收信号的符号率采样。采用增益调度策略进行定时更新;一个带有大增益的短训练符号集用于定时更新,然后是带有低增益的数据符号用于定时更新。这允许在训练期间使用较大的校正步长,从而导致快速收敛。它特别适用于全数字光学系统
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引用次数: 1
Application of object-oriented techniques to the OAM&P of telecommunications networks 面向对象技术在电信网络OAM&P中的应用
P. Johnson
The author examines the recent efforts of the ANSI T1M1.5 working group and T1M1 committee to apply object-oriented (O-O) data modeling techniques to the development of standards for the operation, administration, maintenance, and provisioning (OAM&P) of telecommunications networks (TNs). The use of these state-of-the-art data modeling techniques is launching the North American telecommunications network into an open, standard, multi-vendor communications environment. The author presents a description of the organization of the T1M1 standards body, a brief overview of the Telecommunications Management Network (TMN) model, the basic principles of O-O data modeling, a description of recent network management standardization efforts in T1M1 that use these techniques, and a brief description of current T1M1 work.<>
作者考察了ANSI T1M1.5工作组和T1M1委员会最近将面向对象(O-O)数据建模技术应用于电信网络(tn)的操作、管理、维护和供应(OAM&P)标准开发的工作。使用这些最先进的数据建模技术将北美电信网络推向一个开放的、标准的、多供应商的通信环境。作者描述了T1M1标准机构的组织,简要概述了电信管理网络(TMN)模型,O-O数据建模的基本原则,描述了最近在T1M1中使用这些技术的网络管理标准化工作,并简要描述了当前T1M1工作
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引用次数: 0
Statistical analysis and simulation study of video teleconference traffic in ATM networks ATM网络中视频电话会议流量的统计分析与仿真研究
D. Heyman, A. Tabatabai, T. V. Lakshman
Some of the source modeling and performance issues related to providing video teleconference services over asynchronous transfer mode (ATM) networks were studied. Under certain circumstances, traffic periodicity (due to the constant video frame rate) can cause different sources with identical statistical characteristics to experience cell-loss rates which can differ by several orders of magnitude. Some of this source-periodicity effect can be mitigated by appropriate buffer scheduling. For the video teleconference sequence analyzed (without scene changes or scene cuts and with moderate motion), the number of cells per frame is not normally distributed. Instead, it follows a gamma (or negative binomial) distribution. For traffic studies, an autoregressive model of order 2 and a two-state Markov chain model either underestimate or overestimate the occurrence of frames with a large number of cells, and these frames are a primary factor in determining cell-loss rates. The order-2 autoregressive model, however, fits the data well in a statistical sense. A multistate Markov chain model which can be derived from three traffic parameters (mean, correlation, and variance) is sufficiently accurate for use in traffic studies.<>
研究了在异步传输模式(ATM)网络上提供视频电话会议服务的一些源建模和性能问题。在某些情况下,流量的周期性(由于恒定的视频帧率)可能会导致具有相同统计特征的不同源经历可能相差几个数量级的蜂窝损失率。可以通过适当的缓冲区调度来减轻这种源周期性的影响。对于所分析的视频电话会议序列(没有场景变化或场景剪切,运动适中),每帧的单元数不是正态分布的。相反,它遵循gamma(或负二项)分布。对于交通研究,二阶自回归模型和双态马尔可夫链模型低估或高估了具有大量细胞的帧的出现,而这些帧是决定细胞损失率的主要因素。然而,二阶自回归模型在统计意义上很好地拟合了数据。由三个交通参数(均值、相关和方差)推导出的多状态马尔可夫链模型在交通研究中具有足够的准确性
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引用次数: 486
On the rate-distortion efficiency of subband coding 子带编码的率失真效率研究
T. Fischer
It is shown that, for the subband coding of a wide sense stationary Gaussian source, with the implied separate encoding of the subband signals, the encoding performance is generally inferior to the rate-distortion function of the source. Exceptions are when the source is white or when the filters are ideal (brickwall).<>
研究结果表明,对于广义平稳高斯源的子带编码,如果对子带信号进行隐式分离编码,其编码性能一般不如源的率失真函数。例外情况是当源是白色或当过滤器是理想的(砖墙)。
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引用次数: 28
A data-driven multitone echo canceller 数据驱动的多音回声消除器
J. Cioffi, J. Bingham
A data-driven echo canceller for full-duplex data transmission with multitone modulation is presented. This echo canceller is shown not to be impaired by the eigenvalue-spread problems that are inherent in signal-driven echo cancellers, and to have numerical performance and cancellation range that equals, and in most cases exceeds, that of data-driven echo cancellers used in data transmission with baseband or quadrature amplitude modulation. The method makes use of frequency-domain updating, but with time-domain implementation of the canceller. It introduces no delay into the received signal path.<>
提出了一种多音调制全双工数据传输的数据驱动回波消除器。该回波消除器不会受到信号驱动回波消除器固有的特征值传播问题的影响,并且具有与基带或正交调幅数据传输中使用的数据驱动回波消除器相同的数值性能和消除范围,并且在大多数情况下超过该数值性能和消除范围。该方法利用频域更新,但在时域实现消去。它不会在接收信号路径中引入延迟。
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引用次数: 80
Improvement of ADP-CELP speech coding at 4 kbits/s 4 kb /s ADP-CELP语音编码的改进
M. Akamine, K. Miseki, M. Oshikiri
The authors describe improvements in the ADP-CELP (adaptive-density-pulse code-excited-linear-prediction) coder at 4 kb/s. A fractional delay was introduced to the adaptive codebook. A fast exhaustive search procedure for a uniformly distributed fractional delay was studied for the case where the delay length is shorter than the codevector's dimension. A novel method to determine the long-term predictor's gain was also used. Four methods to find the density pattern of the ADP excitation were investigated. A modified adaptive postfilter was used to improve subjective speech quality.<>
作者描述了ADP-CELP(自适应密度-脉冲编码-激励-线性预测)编码器在4 kb/s的改进。在自适应码本中引入分数阶延迟。研究了延迟长度小于协矢量维数时均匀分布分数阶延迟的快速穷举搜索方法。还使用了一种新的方法来确定长期预测器的增益。研究了四种确定ADP激励密度分布的方法。采用一种改进的自适应后滤波器来提高主观语音质量
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引用次数: 3
An effective algorithm to calculate the distribution of the buffer contents and the packet delay in a multiplexer with bursty sources 一种计算突发源多路复用器中缓冲区内容分布和包延迟的有效算法
B. Steyaert, H. Bruneel
A statistical multiplexer with a finite number of independent and identical bursty traffic sources (users) is considered. The burstiness of the sources is modeled by describing both the active periods (during which a user generates one packet per slot) and the passive periods (during which a user does not generate any data) as geometric random variables. As a result, a correlated-arrivals queuing model for the multiplexer buffer is obtained, and it is analyzed by numerical means. Specifically, algorithms are given for the explicit derivation of the probability mass functions of the buffer contents (in packets) and the packet delay (in slots). The results indicate a very strong influence of the burstiness of the required buffer space in the multiplexer and the possible values of the packet delay, even for a given mean arrival rate per slot.<>
考虑具有有限个独立且相同的突发流量源(用户)的统计复用器。源的突发性是通过将活动周期(在此期间用户每个时隙生成一个数据包)和被动周期(在此期间用户不生成任何数据)描述为几何随机变量来建模的。得到了多路复用器缓冲区的相关到达队列模型,并用数值方法对其进行了分析。具体来说,给出了显式推导缓冲区内容(包中)和包延迟(槽中)的概率质量函数的算法。结果表明,即使对于给定的每个时隙平均到达率,多路复用器中所需缓冲空间的突发性和数据包延迟的可能值也会产生非常强的影响。
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引用次数: 18
Demonstrators and integration 演示和集成
O. Baireuther, R. Vinckier
It is argued that within the framework of the RACE (Research and Development in Advanced Communications in Europe) program, demonstrators and integration activities are essential contributions. They validate the work carried out in areas like strategies and technology and support application pilots. Their validity is applicable for all the elements of telecommunication networks, such as terminals, transmission, switching, network management and administration contained under the program. Various levels of the applied demonstrations are outlined and two integration projects are briefly noted.<>
有人认为,在RACE(欧洲先进通信研究与发展)计划的框架内,演示和集成活动是必不可少的贡献。他们验证在战略和技术等领域开展的工作,并支持应用试点。其有效性适用于程序所包含的终端、传输、交换、网络管理和管理等电信网络的所有要素。本文概述了应用演示的各个层次,并简要说明了两个集成项目。
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引用次数: 0
Integrated access: the achievement of a vision 综合准入:实现一个愿景
Michael J Forbes, S. P. Spencer
The authors highlight many of the open challenges to fulfilment of the technical specification for TR-303. The telephone operating companies have a vision of integrated access that focuses on increased revenues at a reduced cost in a timely fashion. TR-303 is a technical solution that begins to address many of the issues embodied in this vision. While there are many technical challenges, the key to the timely introduction of TR-303 technology is compatibility with the existing investment in operations systems (OSs), local digital switches (LDSs) and digital loop carrier systems. TR-303 will begin with a core feature set that will be proven in close interaction between the LDS and the remote digital terminal. The existence of working TR-303 systems in the network, working with existing OSs, will provide a springboard for new OS functionality as well as providing a stable interface for development.<>
作者强调了实现TR-303技术规范的许多开放挑战。电话运营公司有一个综合接入的愿景,即以及时降低成本的方式增加收入。TR-303是一种技术解决方案,开始解决这一愿景中体现的许多问题。虽然存在许多技术挑战,但及时引入TR-303技术的关键是与现有操作系统(os)、本地数字交换机(lds)和数字环路载波系统的兼容性。TR-303将从一个核心功能集开始,该功能集将在LDS和远程数字终端之间的密切交互中得到验证。网络中工作的TR-303系统的存在,与现有的操作系统一起工作,将为新的操作系统功能提供一个跳板,并为开发提供一个稳定的接口。
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引用次数: 0
New feature extraction methods and the concept of time-warped distance in speech processing 语音处理中新的特征提取方法和时间扭曲距离的概念
G. Gordos
For pitch detection, voiced/unvoiced decisions and speech/nonspeech decisions, an improved average magnitude difference function (AMDF) is described that has given promising results: adaptation improves accuracy and skeletonization speeds up computation. A novel definition of time-warped distance results in decreased error probability in speech recognition; however, no fast algorithm for its computation has yet been found. The concept of time-warped average, on the other hand, is easy to compute and results in better speech recognition score. Both improved AMDF and time-warped distance are discussed for use in the speaker identification environment.<>
对于音高检测、浊音/浊音决策和语音/非语音决策,描述了一种改进的平均幅度差函数(AMDF),该函数给出了有希望的结果:自适应提高了准确性,骨架化加快了计算速度。提出了一种新的时间扭曲距离定义,降低了语音识别中的错误概率;然而,目前还没有找到快速的计算算法。另一方面,时间扭曲平均值的概念易于计算,并且可以获得更好的语音识别分数。讨论了改进的AMDF和时间扭曲距离在说话人识别环境中的应用
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引用次数: 1
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IEEE Global Telecommunications Conference GLOBECOM '91: Countdown to the New Millennium. Conference Record
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