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2005 5th International Conference on Information Communications & Signal Processing最新文献

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Blind Identification of OFDM Signal in Rayleigh Channels 瑞利信道OFDM信号的盲识别
Pub Date : 2005-12-06 DOI: 10.1109/ICICS.2005.1689190
Bin Wang, L. Ge
This paper proposes an algorithm for the blind identification of OFDM (orthogonal frequency division multiplexing) signal from the single-carrier modulated signals in the Rayleigh channels. The algorithm doesn't need any prior knowledge of the signals, such as SNR (signal to noise ratio), symbol rate and carrier frequency, and deals with the bandpass signals directly avoiding the carrier recovering process. Simulation results prove the feasibility and show that an over 95% correct classification rate could be reached when the SNR is above 0 dB
提出了一种从瑞利信道中单载波调制信号中盲识别正交频分复用(OFDM)信号的算法。该算法不需要任何信号的先验知识,如信噪比、符号率和载波频率,直接处理带通信号,避免了载波恢复过程。仿真结果证明了该方法的可行性,当信噪比大于0 dB时,分类正确率可达95%以上
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引用次数: 15
Comparison of Session Establishment Schemes over IMS in Mobile Environment 移动环境下IMS会话建立方案的比较
Pub Date : 2005-12-06 DOI: 10.1109/ICICS.2005.1689125
M. Alam, Z.D. Wu
The existing session establishment scenario of IP multimedia subsystem (IMS) suffers from triangular routing for a certain period of time when an end SIP user or terminal is mobile. In this paper, the performance of three possible session establishment scenarios in a mobile environment is investigated by using an analytical model. The model is developed based on the expressions of cost functions, which may represent system delay and overhead involved in sessions establishment. Some numerical results are demonstrated which provides useful insights into the system behavior. Finally, a decision algorithm is demonstrated in this article to select the optimal session set up option
现有的IP IMS (multimedia subsystem)会话建立场景,在终端SIP用户或终端处于移动状态时,会有一段时间出现三角路由。本文利用分析模型研究了移动环境下三种可能的会话建立场景的性能。该模型是基于代价函数的表达式建立的,它可以表示系统延迟和会话建立所涉及的开销。给出了一些数值结果,为系统行为提供了有用的见解。最后,本文给出了一种选择最优会话设置选项的决策算法
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引用次数: 6
Propagation Path Loss Models for Parking Buildings 停车楼传播路径损耗模型
Pub Date : 2005-12-06 DOI: 10.1109/ICICS.2005.1689276
S. Phaiboon
This paper proposes a method to predict path loss in parking buildings. We performed measurements in two different of parking buildings at a frequency of 1.8 GHz and proposed new upper- and lower-bound models. The actual breakpoint distance was shorter than theoretical one because of the effective floor height due to cars. These models also include time-variant effect, cars in parking area. This time-variant effect influence on the wave propagation very high. The results have shown that the upper- and lower-bound models will be useful for the system and cell design of indoor wireless communication systems
提出了一种停车楼路径损失预测方法。我们在1.8 GHz的频率下对两个不同的停车场进行了测量,并提出了新的上限和下限模型。实际的断点距离比理论的短,因为汽车的有效地板高度。这些模型还包括时变效应、停车区域内的车辆。这种时变效应对波的传播影响很大。结果表明,所建立的上界和下界模型对室内无线通信系统的系统和小区设计具有实用价值
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引用次数: 16
On A Real-valued EVD Approach to Cyclic DOA Estimation 循环DOA估计的一种实值EVD方法
Pub Date : 2005-12-06 DOI: 10.1109/ICICS.2005.1689263
Zhigang Liu, Jinkuan Wang, Yanbo Xue
By exploiting the real-valued eigen decomposition (EVD) of the centro-Hermitian matrix, a novel cyclic MUSIC algorithm with signal selective property is presented by introducing a new forward backward smoothed covariance matrix. Compared with cyclic MUSIC algorithm, the proposed approach has a better performance in the presence of multipath propagation. In addition, this approach not only reduces the computational complexity, but also allows to select desired signals and to ignore interferences by exploiting the cyclostationarity property of signals of interest (SOIs). Simulation results that illustrate the performance of this approach in conjunction with cyclic MUSIC algorithm are described
利用中心厄米矩阵的实值特征分解(EVD),通过引入一种新的前向后平滑协方差矩阵,提出了一种具有信号选择性的循环MUSIC算法。与循环MUSIC算法相比,该方法在存在多径传播的情况下具有更好的性能。此外,该方法不仅降低了计算复杂度,而且还允许选择所需信号,并通过利用感兴趣信号(SOIs)的循环平稳性忽略干扰。仿真结果说明了该方法结合循环MUSIC算法的性能
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引用次数: 0
Variable Step-Size Pradhan-Reddy Subband Adaptive Filters 可变步长Pradhan-Reddy子带自适应滤波器
Pub Date : 2005-12-06 DOI: 10.1109/ICICS.2005.1689181
M. Abadi, J. H. Husøy
Employing a recently introduced framework within which a large number of classical and modern adaptive filter algorithms can be viewed as special cases, we develop a generic variable step size adaptive filter. Variable step-size (VSS) least mean square (VSSLMS), VSS normalized LMS (VSSNLS) and VSS affine projection algorithms (VSSAPA) are particular examples of adaptive algorithms covered by this generic adaptive filter. In this paper we introduce the new VSS subband adaptive filter referred to as the VSS Pradhan-Reddy subband adaptive filter (VSSPRSAF) algorithm. The proposed algorithm exhibits fast convergence and lower steady state mean square error when compared to the ordinary PRSAF algorithm
采用最近引入的框架,其中大量经典和现代自适应滤波器算法可以被视为特殊情况,我们开发了一个通用的可变步长自适应滤波器。可变步长(VSS)最小均方(VSSLMS), VSS归一化LMS (VSSNLS)和VSS仿射投影算法(VSSAPA)是该通用自适应滤波器涵盖的自适应算法的特殊示例。本文介绍了一种新的VSS子带自适应滤波器,即VSS Pradhan-Reddy子带自适应滤波器(VSSPRSAF)算法。与普通PRSAF算法相比,该算法收敛速度快,稳态均方误差小
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引用次数: 3
Comparison of the Efficiency of Ordered and Disordered Codebook Techniques in Speech Coding 语音编码中有序和无序码本技术效率的比较
Pub Date : 2005-12-06 DOI: 10.1109/ICICS.2005.1689033
J. Srinonchat
Many coding techniques have been proposed for the compression of speech signals. Speech signals are continuous signals shaped by the movement of the vocal tract. Linear predictive coefficients (LPC) and line spectral pairs (LSP) coefficients have been used to model the position of the vocal tract. Each speaker's vocal tract uses a unique, finite set of positions to generate their voice. This paper compares of the performance of ordered and disordered codebooks which employed to measure and classify the repetition of the speech signal coefficients. An analysis and results of the efficient performance of both types of codebooks in terms of the quality of synthesised speech are also provided. Also both ordered and disordered codebooks can reduce the number of bit rate transmission around 20% in speech coding system
针对语音信号的压缩,已经提出了许多编码技术。语音信号是由声道运动形成的连续信号。线性预测系数(LPC)和线谱对(LSP)系数已被用来模拟声道的位置。每个说话者的声道都使用一组独特的、有限的位置来发出声音。本文比较了用有序码本和无序码本对语音信号重复系数进行测量和分类的性能。本文还提供了两种类型的码本在合成语音质量方面的有效性能分析和结果。在语音编码系统中,有序码本和无序码本都可以减少20%左右的比特率传输
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引用次数: 3
Cheannel Capacity of Twisted Wire Pairs in Multi-Pair Symmetric Copper Cables 多对对称铜缆中双绞线对的通道容量
Pub Date : 2005-12-06 DOI: 10.1109/ICICS.2005.1689215
Christoph Lange, Andreas Ahrens
Crosstalk is caused by electromagnetic coupling between neighbouring wire pairs in multi-pair copper cables. In this contribution the channel capacity of a twisted wire pair in an unidirectional driven multi-pair cable is calculated, where only far-end crosstalk is present. Thereby two scenarios are considered: Firstly the far-end crosstalk from neighbouring wire pairs is interpreted as disturbance, which affects the transmission in the considered wire pair. Secondly the far-end crosstalk paths are viewed as additional transmission paths, which together with the wanted signal path convey the signal from the near to the far cable end. In the former case the channel capacity is decreased compared to the transmission with only noise disturbance, whereas in the latter case the channel capacity is increased due to the utilization of additional transmission paths. In particular for short cables the effect of far-end crosstalk is very strong and with rising number of neighbouring wire pairs the far-end crosstalk impact increases
串扰是由多对铜缆中相邻导线对之间的电磁耦合引起的。在此贡献中,计算了单向驱动多对电缆中双绞线对的信道容量,其中仅存在远端串扰。因此,考虑了两种情况:首先,来自相邻线对的远端串扰被解释为干扰,它影响在所考虑的线对中的传输。其次,将远端串扰路径视为附加传输路径,它与所需信号路径一起将信号从近端传输到远端。在前一种情况下,与仅具有噪声干扰的传输相比,信道容量减少,而在后一种情况下,由于使用了额外的传输路径,信道容量增加。特别是对于短电缆,远端串扰的影响非常强,并且随着相邻导线对数量的增加,远端串扰的影响也会增加
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引用次数: 2
A Fast Handoff Scheme for Mobile IPv6 Based Wireless Networks 基于移动IPv6无线网络的快速切换方案
Pub Date : 2005-12-06 DOI: 10.1109/ICICS.2005.1689131
P. Kim, E. Kim, E. Lee
This paper proposes a new fast handoff scheme for Mobile IPv6 based wireless networks. To implement the proposed scheme, the beacon message used in L2 is defined newly by adding a specific subfield to the capability information field. In addition, Router Information Request/Reply messages used in L3 are defined newly. The proposed scheme provides the faster acquisition of new access router's network information than the existing one, which might be advantageous for the low handoff latency. In addition, the proposed scheme might reduce amount of traffic in comparison with the existing one, which might be remarkable when there are many mobile nodes that are now connecting to current access router. Moreover, in the proposed scheme, a mobile node can know whether it changes access router or access point, which can remove redundant traffic of the existing one. Via extensive experiments, it will be shown that the proposed scheme outperforms existing ones in terms of the L3 handoff latency and the packet loss
提出了一种新的基于移动IPv6的无线网络快速切换方案。为了实现所提出的方案,通过在能力信息字段中添加特定的子字段来重新定义L2中使用的信标消息。另外,重新定义了L3使用的Router Information Request/Reply报文。该方案能够比现有方案更快地获取新的接入路由器的网络信息,这可能有利于低切换延迟。此外,与现有方案相比,所提出的方案可能会减少通信量,当有许多移动节点现在连接到当前的接入路由器时,这可能是显着的。此外,在该方案中,移动节点可以知道它是否要更换接入路由器或接入点,从而可以消除现有接入点的冗余流量。通过大量的实验,将证明所提出的方案在L3切换延迟和丢包方面优于现有方案
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引用次数: 0
Support System for Telecommunication Using Keywords 关键词:电信支持系统
Pub Date : 2005-12-06 DOI: 10.1109/ICICS.2005.1689220
K. Asai, F. Saito
We developed a prototype system to support telecommunications that uses keywords selected by the presenter in videoconference. The presenter holds to show keyword cards, and the listeners at remote sites can see additional information with the keywords. Although keyword captions are considered effective in video learning environments for learning foreign languages, we think they should also be available for interactive communications. Our prototype system recognizes letters and characters in a video image, and provides us with additional functions, such as language translation, 3D models, and audio reproductions. The visual data are overlaid onto the real scene with a multilayered display, using a Web browser as a presentation tool to enable us to easily author/edit multimedia data. Optical character reader (OCR) middleware was implemented into the recognition function for Japanese, and an open-source library, ARToolkit was used to detect the character markers that controlled the functions of the information presentation. The support system is designed to efficiently exploit audiovisual signals of existing videoconferencing systems
我们开发了一个支持电信的原型系统,该系统在视频会议中使用演示者选择的关键字。演示者坚持展示关键字卡片,远程站点的听众可以看到关键字的附加信息。虽然关键字字幕被认为是在视频学习环境中学习外语的有效方法,但我们认为它们也应该用于互动交流。我们的原型系统可以识别视频图像中的字母和字符,并为我们提供额外的功能,如语言翻译、3D模型和音频复制。可视化数据通过多层显示叠加在真实场景上,使用Web浏览器作为表示工具,使我们能够轻松地编写/编辑多媒体数据。将光学字符阅读器(OCR)中间件实现到日语识别功能中,并利用开源库ARToolkit检测控制信息表示功能的字符标记。该支持系统旨在有效地利用现有视频会议系统的视听信号
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引用次数: 0
A New Adaptive Filter Structure for Tracking Multiple Sinusoids 一种新的多正弦跟踪自适应滤波器结构
Pub Date : 2005-12-06 DOI: 10.1109/ICICS.2005.1689071
Hanwu Sun, L. Shue, Jiangfeng Chen
This paper presents a new cascade filtering structure using feedback signals for tracking the frequencies of multiple sinusoids. Its chief advantage lies in computational savings over earlier structures of this type. Simulation results show that this streamlined structure can also efficiently overcome "tracking ripple", delivering very good steady-state tracking
本文提出了一种利用反馈信号跟踪多个正弦波频率的新型级联滤波结构。它的主要优点在于与这种类型的早期结构相比节省了计算量。仿真结果表明,这种流线型结构还可以有效地克服“跟踪纹波”,提供非常好的稳态跟踪
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引用次数: 0
期刊
2005 5th International Conference on Information Communications & Signal Processing
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