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EUROCON'2001. International Conference on Trends in Communications. Technical Program, Proceedings (Cat. No.01EX439)最新文献

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Technology and democracy: a panacea for government? 科技和民主:政府的灵丹妙药?
G. Alsop, C. Wills
This paper considers whether communications theory can help us to understand better the social and juridical impacts of new telecommunications services, particularly in relation to e-democracy and e-government.
本文考虑通信理论是否可以帮助我们更好地理解新的电信服务的社会和法律影响,特别是与电子民主和电子政府有关的影响。
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引用次数: 0
Formal definition of SIP end systems behavior SIP终端系统行为的正式定义
R. Radovic, I. crkvenac, S. Srbljic
The rapid improvement of hardware performance and the constantly growing bandwidth of computers have encouraged the development of communications based on computer networks. The first step in this direction is the definition of a set of protocols that will enable the design and implementation of various multimedia communication services. SIP (session initiation protocol) is a protocol for initiating multimedia conferences, and is a proposed standard, but its behavior is still not formally defined. We give a formal definition of the behavior of SIP end systems in the form of automata that include all features currently supported by SIP. The automata are defined using UML. This greatly simplifies the implementation of SIP systems compatibility between different systems based on SIP, because there are software tools that can translate UML specifications directly into the high-level languages (e.g., C++, Java).
硬件性能的快速提高和计算机带宽的不断增长促进了基于计算机网络的通信的发展。这个方向的第一步是定义一组协议,使设计和实现各种多媒体通信服务成为可能。SIP(会话发起协议)是一种用于发起多媒体会议的协议,是一种被提议的标准,但其行为还没有正式定义。我们以自动机的形式给出了SIP端系统行为的正式定义,包括SIP当前支持的所有特性。自动机是使用UML定义的。这极大地简化了基于SIP的不同系统之间SIP系统兼容性的实现,因为有软件工具可以将UML规范直接转换为高级语言(例如,c++, Java)。
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引用次数: 8
Find-reassemble-path algorithm for finding node disjoint paths in telecommunications networks with two technologies 基于两种技术的电信网络节点不相交路径查找-重组路径算法
Kwang-Fu Li, Yueh-Hsia Tsai, Chia-Ching Li
In this paper, we present a new algorithm, the find-reassemble-path algorithm, for finding two node disjoint paths with lower cost than the original heuristics. We use the approach that reverses the directions of the arcs of the first shortest path, instead of deleting all the nodes on the path and all the arcs incident to the path, and then find second paths on the new graph. Finally, we use these two paths to reassemble into two node disjoint paths. There are four major steps in this algorithm. The first two steps are used to find paths and the last two steps are used to reassemble paths.
在本文中,我们提出了一种新的算法,即寻找-重组-路径算法,用于寻找两个节点不相交的路径,其成本比原来的启发式算法低。我们使用的方法是反转第一条最短路径的弧线方向,而不是删除路径上的所有节点和与路径相关的所有弧线,然后在新图上找到第二条路径。最后,我们将这两条路径重新组合成两条节点不相交的路径。这个算法有四个主要步骤。前两步用于查找路径,后两步用于重新组装路径。
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引用次数: 1
New cepstral zero-pole vocal tract models for TTS synthesis 新的背侧零极声道模型用于TTS合成
R. Vích, J. Pribil, Z. Smekan
Speech is an analog sound signal produced by exciting the human vocal tract. The magnitude response of the vocal tract exhibits both peaks (formants) and valleys (antiformants). Vocal tract models are differentiated according to whether they model the formants alone (LPC models) or also antiformants (ARMA and cepstral models). New structures are proposed for an effective realization of cepstral vocal tract models that model both formants and antiformants.
语音是通过刺激人的声道而产生的模拟声音信号。声道的幅度响应呈现峰(共振峰)和谷(反共振峰)。根据声道模型是单独模拟共振峰(LPC模型)还是同时模拟反共振峰(ARMA和背侧模型)来区分声道模型。提出了一种新的结构来有效地实现背侧声道模型,该模型可以同时模拟共振峰和反共振峰。
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引用次数: 19
'Errors-and-erasures' decoding in RS coded DS-CDMA using noncoherent M-ary orthogonal modulation RS编码DS-CDMA中使用非相干正交调制的“错误和擦除”译码
Lie-liang Yang, L. Hanzo
In this contribution, we investigate the performance of Reed-Solomon (RS) coded DS-CDMA using noncoherent M-ary orthogonal modulation over multipath Rayleigh fading channels. 'Errors-and-erasures' RS decoding is considered, where erasures are declared based on various erasure insertion schemes, such as the maximum output threshold test (OTT), ratio threshold test (RTT) and the joint maximum output and ratio threshold test (MOR-TT). Furthermore, the performance of 'errors-and-erasures' decoding employing the joint MOR-TT is compared to that of 'error-correction-only' decoding and to that of 'errors-and-erasures' decoding using the OTT as well as the RTT. The numerical results show that, when using 'errors-and-erasures' decoding, RS codes of a given code rate can achieve a significantly higher coding gain than without erasure information, and that the MOR-TT technique outperforms both the RTT and the OTT techniques.
在这篇文章中,我们研究了在多径瑞利衰落信道上使用非相干正交调制的里德-所罗门(RS)编码的DS-CDMA的性能。考虑了“错误和擦除”RS解码,其中擦除是基于各种擦除插入方案声明的,例如最大输出阈值测试(OTT),比率阈值测试(RTT)和联合最大输出和比率阈值测试(more - tt)。此外,将采用联合MOR-TT的“错误和擦除”解码的性能与“仅纠错”解码的性能以及使用OTT和RTT的“错误和擦除”解码的性能进行了比较。数值结果表明,当使用“错误-擦除”解码时,给定码率的RS码比不使用擦除信息的RS码可以获得更高的编码增益,并且mort - tt技术优于RTT和OTT技术。
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引用次数: 1
A real-time DSP blind signal separation experimental system based on a new simplified mixing model 基于新型简化混频模型的实时DSP盲信号分离实验系统
Peiyu He, P. Sommen, B. Yin
A real-time blind signal separation experimental system has been designed and implemented using TMS320C40 DSP and LabVIEW virtual instruments software. The system is based on a new simplified mixing model that we proposed recently. The simplified model uses the fact that the acoustic transfer functions from a single source to two closely spaced (approximately several 10 cm apart from each other) microphones are very similar. Only the difference between these two transfer functions is needed. The main function of this DSP system is using the measured difference between two impulse responses (DIR) from one source to two microphones to realize the real-time blind signal separation. The real-time experimental results have shown that the proposed simplified mixing model is feasible. In particular, the blind separation performance can be improved considerably by introducing a time delay to solve the inverse problem of a non-minimum phase transfer function.
利用TMS320C40 DSP和LabVIEW虚拟仪器软件,设计并实现了实时盲信号分离实验系统。该系统基于我们最近提出的一种新的简化混合模型。简化模型利用了这样一个事实,即从单个声源到两个紧密间隔(彼此相距约10厘米)的麦克风的声传递函数非常相似。只需要这两个传递函数的差。该DSP系统的主要功能是利用测量到的一个信号源对两个传声器的脉冲响应差(DIR)来实现实时盲信号分离。实时实验结果表明,所提出的简化混合模型是可行的。特别是,通过引入时间延迟来解决非最小相传递函数的逆问题,可以大大提高盲分离性能。
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引用次数: 6
Millimetre wave utilisation and propagation research in the Czech Republic 捷克共和国毫米波的利用和传播研究
V. Kvicera
TESTCOM is engaged in an experimental propagation research in the frequency bands over 10 GHz. The research is especially focused on attenuation due to hydrometeors on line-of-sight paths. The aim of our research is to obtain long-term radio and radio meteorological data in our climatic conditions and to use them for proposals of cumulative distributions of attenuation due to hydrometeors for the average year and the average worst month. These distributions are applied to the calculations of quality and reliability of terrestrial fixed systems.
TESTCOM从事10ghz以上频段的实验传播研究。研究的重点是由于水成物在视线路径上的衰减。我们的研究目的是在我们的气候条件下获得长期的无线电和无线电气象资料,并利用这些资料提出平均年和平均最恶劣月份的水文衰减累积分布建议。这些分布用于计算地面固定系统的质量和可靠性。
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引用次数: 1
A novel design for DCSK communication system 一种新的DCSK通信系统设计
Liu Hu, Z. Jing, L. Bo
In a DCSK (differential chaos shift keying) system implemented with noise suppression, there are three sources of variance: the DCSK signal itself, additive channel noise and the noise suppression scheme. In this paper, a novel DCSK scheme with chaotic polyphase sequence (CPS) is presented to avoid the variance of the DCSK signal itself and decrease the variance for additive channel noise. The simulation results show that the noise performance of the novel DCSK system is slightly degraded in contrast to FM-DCSK for the estimation variance of the noise suppression scheme. However, this design can be implemented easily.
在具有噪声抑制的DCSK(微分混沌移位键控)系统中,有三个方差源:DCSK信号本身、加性信道噪声和噪声抑制方案。为了避免DCSK信号本身的方差,减小加性信道噪声的方差,提出了一种新的混沌多相序列(CPS) DCSK方案。仿真结果表明,在噪声抑制方案的估计方差方面,与FM-DCSK相比,新型DCSK系统的噪声性能略有下降。然而,这种设计可以很容易地实现。
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引用次数: 1
DTEx-Sys-a Web oriented intelligent tutoring system dtex - sys -面向Web的智能辅导系统
M. Rosić, V. Glavinic, S. Stankov
Distance learning systems (DLSs) are a class of important services that information infrastructures have to provide users. DLSs base their operation on information access, electronic transaction and communication services furnished by information infrastructures. Regarding the teaching process, by using DLSs new paradigms enable users to distribute educational content to support interaction among user classes (instructor-student, student-student and student-educational institution), testing, evaluation, and eventually advice in different domain knowledge areas. Classification of DLSs depends on the way these paradigms are implemented. In this paper, we concentrate on a particular DLS class, which encompass systems based on enabling Web access to otherwise classical intelligent tutoring systems-Web oriented intelligent tutoring systems. In this paper we present one such system-distributed tutor-expert system (DTEx-Sys), which has been developed for asynchronous distance educational purposes. DTEx-Sys is designed as a 3-tier client-server architecture where the intelligent tutoring functions are separated from the user interface and the knowledge base(s). The system functionality comprehends knowledge base access for arbitrary domain knowledge, along with testing, diagnosing and evaluation of students' work.
远程学习系统(dls)是信息基础设施必须向用户提供的一类重要服务。服务供应商的运作是基于资讯基建所提供的资讯存取、电子交易及通讯服务。在教学过程方面,通过使用dls,新的范例使用户能够分发教育内容,以支持用户类(教师-学生、学生-学生和学生-教育机构)之间的交互、测试、评估以及不同领域知识领域的最终建议。dls的分类取决于这些范例的实现方式。在本文中,我们专注于一个特定的DLS类,它包含了基于Web访问经典智能辅导系统的系统-面向Web的智能辅导系统。本文提出了一种为异步远程教育而开发的分布式教师专家系统(DTEx-Sys)。DTEx-Sys采用三层客户端-服务器架构,将智能辅导功能与用户界面和知识库分离。系统功能包括对任意领域知识的知识库访问,以及对学生作业的测试、诊断和评价。
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引用次数: 6
Packet based telephony 分组电话
B. Vlaovič, Z. Brezočnik
This paper describes methods used to place telephone calls over packet-based networks with emphasis on Internet point-to-point communication. Two approaches will be described-H.323 and SIP (session initiation protocol). The first is an ITU standard and is already widely deployed by commercial vendors. The second, SIP, was developed by IETF activists and represents an "Internet approach" to telephony. First, a brief overview of reasons for the growing importance of IP telephony is given. It is followed by a short introduction to voice transmission over packet networks and an overview of the existing standards. In the second part of the paper, short examples of H.323 and SIP call setup are given.
本文描述了用于在基于分组的网络上放置电话呼叫的方法,重点是Internet点对点通信。本文将描述两种方法。323和SIP(会话发起协议)。第一种是国际电联的标准,已经被商业供应商广泛部署。第二个SIP是由IETF积极分子开发的,代表了电话的“互联网方法”。首先,简要概述了IP电话日益重要的原因。接下来是对分组网络上的语音传输的简短介绍和对现有标准的概述。第二部分给出了H.323协议和SIP协议呼叫建立的实例。
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引用次数: 9
期刊
EUROCON'2001. International Conference on Trends in Communications. Technical Program, Proceedings (Cat. No.01EX439)
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