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Seventh IEEE International Symposium on Multimedia (ISM'05)最新文献

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Distributed user interface elements to support smart interaction spaces 分布式用户界面元素,支持智能交互空间
Pub Date : 2005-12-12 DOI: 10.1109/ISM.2005.52
K. Luyten, K. Coninx
In this paper we show how an interactive system can be distributed among several peer devices. By taking advantage of the current trend towards ambient intelligent environments, we can make use of a combination of computing resources in the surrounding of the user to function as one logical interactive system; an interaction space. Our approach relies on the fact that nowadays most computing resources are network enabled and publish their device profile using some special purpose protocol. For this reason, federations of devices that support the tasks of the user can be composed automatically according to the requirements of these tasks. This distribution of the user interface over a federation of devices can be local or nonlocal. It raises the opportunity for supporting collaborative tasks with the same user interface with little or no extra effort from the user interface designer. Future tools supporting the design, creation and deployment of distributed interactive systems using device federations should maintain usability and usefulness of a dynamic distributed system. We use two different metrics to cope with these problems: interface completeness and interface continuity.
在本文中,我们展示了如何将一个交互系统分布在几个对等设备之间。通过利用当前环境智能环境的趋势,我们可以利用用户周围的计算资源组合作为一个逻辑交互系统;一个交互空间。我们的方法依赖于这样一个事实,即现在大多数计算资源都是网络支持的,并且使用一些特殊用途的协议发布它们的设备配置文件。因此,支持用户任务的设备联盟可以根据这些任务的要求自动组成。用户界面在联合设备上的这种分布可以是本地的,也可以是非本地的。它为使用相同的用户界面支持协作任务提供了机会,而用户界面设计人员很少或没有额外的工作。支持使用设备联合的分布式交互系统的设计、创建和部署的未来工具应该保持动态分布式系统的可用性和有用性。我们使用两个不同的度量来处理这些问题:接口完备性和接口连续性。
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引用次数: 77
The protection of QoS for multimedia transmission against denial of service attacks 针对拒绝服务攻击的多媒体传输QoS保护
Pub Date : 2005-12-12 DOI: 10.1109/ISM.2005.115
Hongli Luo, M. Shyu
In this paper, a secure and adaptive multimedia transmission framework is proposed to maintain the quality of service (QoS) of the multimedia streams during the denial-of service (DoS) attacks. A DoS attack generates a large amount of traffic, which will occupy the bandwidth and degrade the quality of video. The proposed framework consists of two components: intrusion detection and adaptive transmission management. The intrusion detection component monitors the incoming traffic to the server, detects the attacks, and interacts with the adaptive transmission management component; while the adaptive transmission management component is designed to improve QoS of the video via the efficient utilization of the network resources. With the detection of the DoS attacks, the bandwidth occupied by the attacks can be reduced and protected for video transmission. The results of our preliminary simulations in NS2 show that the quality of the multimedia stream can still be maintained during an attack.
本文提出了一种安全的、自适应的多媒体传输框架,以保证多媒体流在遭受拒绝服务攻击时的服务质量。DoS攻击会产生大量流量,占用带宽,降低视频质量。该框架由入侵检测和自适应传输管理两部分组成。入侵检测组件监视进入服务器的流量,检测攻击,并与自适应传输管理组件交互;自适应传输管理组件通过对网络资源的有效利用来提高视频的QoS。通过检测DoS攻击,可以减少攻击占用的带宽,保护视频传输。我们在NS2中的初步仿真结果表明,在攻击过程中仍然可以保持多媒体流的质量。
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引用次数: 5
Larynx virtual surgery 喉虚拟手术
Pub Date : 2005-12-12 DOI: 10.1109/ISM.2005.74
A. Montagnoli, E. Fonseca
This paper presents a novel computational method designed to assist phonologists to anticipate the effects produced in the voice, as a result of physical and mechanical alterations of the larynx model, due to a surgery to correct dysphonia. The technique is based on images obtained from endoscopic exams of the larynx. The main objective of the study was to use the active contours method to develop a larynx model based on vocal fold and glottis movements. Starting with the recorded voice signal of subject, we filter the characteristics of the glottal pulse obtained from the image exam, assuming that the dysphonia is caused by irregularities of the larynx. The modifications made in the glottis model are used to estimate a new glottal filter, which is then added to the filtered signal. Using this method, one can obtain a new voice that maintains the individual's personal characteristics after virtual surgery modifications.
本文提出了一种新的计算方法,旨在帮助音系学家预测由于矫正发音障碍的手术导致喉部模型的物理和机械改变而产生的声音效果。该技术是基于从喉部内窥镜检查中获得的图像。本研究的主要目的是利用活动轮廓法建立一个基于声带和声门运动的喉部模型。从被试录制的语音信号开始,我们对图像检查得到的声门脉冲特征进行滤波,假设发声障碍是由喉部不规则引起的。在声门模型中所做的修改用于估计一个新的声门滤波器,然后将其添加到过滤后的信号中。使用这种方法,人们可以在虚拟手术修改后获得保持个人个人特征的新声音。
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引用次数: 2
Consideration of UMTS-WLAN seamless handover UMTS-WLAN无缝切换的考虑
Pub Date : 2005-12-12 DOI: 10.1109/ISM.2005.45
Hye-yeon Kwon, Kwang-Ryul Jung, A. Park, Jaecheol Ryou
By mutual complemented reason of wide coverage and high data transmission, the integration and handover of SG network and WLAN is regarded as an important function for 3G beyond (B3G) mobile communication systems. The seamless mobility is one of the basic requirements for handover between these heterogeneous networks. In this paper, we present a seamless handover mechanism for UMTS and WLAN within the framework of handover architecture, IP mobility and authentication mechanism. Firstly, we review the 3GPP-specific integration architecture of UMTS and WLAN, WLAN UE-initiated tunnel mechanism, IP mobility and authentication mechanisms of both networks. Then we propose the configuration of mobile IPv6 based UMTS-WLAN overlay network and the security mechanism for reducing the handover delay. Lastly we show the operation of seamless handover in our proposed network model.
由于广覆盖和高数据传输的相互补充,SG网络与WLAN的融合和切换被认为是3G超(B3G)移动通信系统的一项重要功能。无缝移动性是异构网络间切换的基本要求之一。本文在切换架构、IP移动性和认证机制的框架下,提出了UMTS和WLAN的无缝切换机制。首先,我们回顾了UMTS和WLAN的3gpp特定集成架构,WLAN的ue发起隧道机制,两个网络的IP移动性和认证机制。然后提出了基于移动IPv6的UMTS-WLAN覆盖网络的配置和减少切换延迟的安全机制。最后给出了该网络模型的无缝切换操作。
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引用次数: 11
Image based tracking of news stories 基于图像的新闻故事跟踪
Pub Date : 2005-12-12 DOI: 10.1109/ISM.2005.67
Jedrzej Z. Miadowicz, J. Gauch, Abhishek Shivadas
Broadcasters often reuse video clips while reporting news stories. New clips are added to the story as it develops replacing old clips. A subset of these video clips may be replayed as follow up stories are investigated. This pattern of video repetition can be used to impose a story level organization on video clips that are broadcast by one or more news channels. In this paper, we describe an automated system for detecting and tracking repeated video clips in news broadcasts. We begin by performing temporal video segmentation to divide the video into shots and scenes. As each frame of the video source is processed, we extract low-level video features that are used to perform repeated sequence detection in realtime. Our matching algorithms have been adapted to recognize partial clip reuse while remaining robust to minor variations in the video source. Our system then builds a set of shots relevant to the news story being tracked, called a story core, and identifies new story episodes by detecting occurrences of shots belonging to this set. The story core is allowed to evolve dynamically over time to account for additional video footage as it becomes available. This technique was evaluated on a 24-hour broadcast of CNN News channel where we were very successful locating and tracking video clips associated with selected news stories.
广播公司在报道新闻时经常重复使用视频片段。随着故事的发展,新的片段被添加到故事中,取代旧的片段。这些视频片段的一个子集可能会在调查后续故事时重播。这种视频重复的模式可以用来对由一个或多个新闻频道播放的视频片段施加故事级别的组织。在本文中,我们描述了一个自动检测和跟踪新闻广播中重复视频片段的系统。我们首先执行时间视频分割,将视频划分为镜头和场景。在处理视频源的每一帧时,我们提取低级视频特征,用于实时执行重复序列检测。我们的匹配算法已经适应了识别部分剪辑重用,同时对视频源中的微小变化保持鲁棒性。然后,我们的系统建立一组与正在跟踪的新闻故事相关的镜头,称为故事核心,并通过检测属于该集的镜头的出现来识别新的故事情节。故事核心可以随着时间的推移而动态发展,以解释额外的视频片段,因为它变得可用。这项技术在CNN新闻频道的24小时广播中进行了评估,我们非常成功地定位和跟踪了与选定新闻故事相关的视频剪辑。
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引用次数: 1
Application of autoregressive decomposition and pole tracking to pathological voice signals 自回归分解和极点跟踪在病理语音信号中的应用
Pub Date : 2005-12-12 DOI: 10.1109/ISM.2005.35
P. Scalassara, M. E. Dajer, Carlos Dias Maciel
In this paper, it is presented an insight of the effects of the application of autoregressive (AR) decomposition and pole tracking to voice signals. The AR model is used to decompose the signals in a set of poles which has a correspondence to the peaks of the signals power spectral density function (PSD). The aim of this work is to show the differences in the behavior of these poles for voice signals collected from two groups of people, one with healthy glottal tract and another with nodule pathology in vocal folds.
本文介绍了自回归(AR)分解和极点跟踪在语音信号中的应用效果。利用AR模型将信号分解为与信号功率谱密度函数(PSD)峰值对应的一组极点。这项工作的目的是从两组人收集的声音信号中显示这些极点的行为差异,一组是健康的声门道,另一组是声带结节病理。
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引用次数: 4
Denial-of-service resistant multicast authentication protocol with prediction hashing and one-way key chain 具有预测哈希和单向密钥链的抗拒绝服务组播认证协议
Pub Date : 2005-12-12 DOI: 10.1109/ISM.2005.47
Seonho Choi
Requirements that need to be met by multicast authentication schemes are packet authenticity, packet loss robustness, loss model independence, and denial of service (DoS) resistance. We develop an efficient multicast authentication scheme for real-time streaming applications that is resistant to denial-of-service attacks while consuming much less resources (CPU and buffer) at receivers compared to previously proposed schemes. This scheme utilizes prediction hashing (PH) and one-way key chain (OKC) techniques based on erasure codes and distillation codes. PH and OKC techniques enable the receiver to significantly reduce the CPU overhead and buffer requirements compared to other block-based solution approaches. Analysis conducted indicates that this new scheme consumes much less CPU time and buffer space than one of the recently proposed denial-of-service (DoS) resistant multicast authentication schemes, pollution resistant authenticated block streams (PRABS) (Karlof et al., 2004), by a factor of more than 5 for buffer requirement and 3 for CPU requirement.
组播认证方案需要满足的要求是报文真实性、丢包鲁棒性、丢失模型无关性和抗DoS (denial of service)。我们为实时流应用开发了一种高效的多播身份验证方案,该方案可以抵抗拒绝服务攻击,同时与之前提出的方案相比,在接收器上消耗更少的资源(CPU和缓冲区)。该方案利用基于擦除码和蒸馏码的预测哈希(PH)和单向密钥链(OKC)技术。与其他基于块的解决方案方法相比,PH和OKC技术使接收器能够显著降低CPU开销和缓冲区需求。分析表明,与最近提出的抗拒绝服务(DoS)组播认证方案——抗污染认证块流(PRABS) (Karlof et al., 2004)相比,这种新方案消耗的CPU时间和缓冲空间要少得多,缓冲需求是5倍以上,CPU需求是3倍以上。
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引用次数: 17
Supporting low-cost video-on-demand in heterogeneous peer-to-peer networks 在异构点对点网络中支持低成本视频点播
Pub Date : 2005-12-12 DOI: 10.1109/ISM.2005.112
Xin Liu, S. Vuong
We propose a novel low-cost video-on-demand architecture for heterogeneous peer-to-peer networks. The key idea of this architecture is to aggregate peers' storage and bandwidths to facilitate on-demand video streaming. To achieve this goal, we split published videos into segments and distribute them to different peers. When watching a video, a peer searches the corresponding segments, and then aggregates bandwidths from multiple supplying peers to stream the video. Instead of relying on the powerful servers/proxies, this architecture exploits the often-underutilized peers' resources, which makes it cost-effective. We demonstrate the effectiveness of this proposed architecture through extensive simulation experiments on large, Internet-like topologies.
我们提出了一种新的低成本视频点播架构,用于异构点对点网络。该体系结构的关键思想是聚合对等体的存储和带宽,以促进点播视频流。为了实现这一目标,我们将发布的视频分成几个片段,并将它们分发给不同的同行。当观看视频时,对等端搜索相应的片段,然后从多个提供的对等端聚合带宽进行视频流传输。这种体系结构不依赖于功能强大的服务器/代理,而是利用通常未充分利用的对等资源,这使得它具有成本效益。我们通过在大型类似互联网的拓扑结构上进行广泛的模拟实验,证明了该架构的有效性。
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引用次数: 13
Hybrid bitrate/PSNR control for H.264 video streaming to roaming users 混合比特率/PSNR控制H.264视频流到漫游用户
Pub Date : 2005-12-12 DOI: 10.1109/ISM.2005.66
F. D. Vito, F. Ridolfo, Juan Carlos De Martin
In wireless communications, the available throughput depends on several parameters, like physical layer, base station distance, fading and interference. Users experience changes in bandwidth within a cell and among same-technology cells, but also among different networks. Moreover, in case of video transmission, the user may specify a desired quality level. The source should encode the stream at a quality as close as possible to this value, without exceeding the available bitrate. We propose a technique to decide whether to encode at constant quality, if resources are enough, or at constant bitrate, if the throughput is not sufficient. With negligible complexity, it proved to obtain better PSNR/bitrate ratios, with respect to only constant bitrate and only constant PSNR coding. We also show this algorithm working in a realistic scenario of a user roaming among heterogeneous networks (WLAN and UMTS). Also in this case, the algorithm proved to achieve high quality/bitrate ratios.
在无线通信中,可用的吞吐量取决于几个参数,如物理层、基站距离、衰落和干扰。用户在一个小区内、同一技术小区之间以及不同网络之间都会经历带宽的变化。此外,在视频传输的情况下,用户可以指定所需的质量水平。源应该在不超过可用比特率的情况下,以尽可能接近该值的质量对流进行编码。我们提出了一种技术来决定是在资源足够的情况下以恒定质量编码,还是在吞吐量不够的情况下以恒定比特率编码。在复杂度可以忽略不计的情况下,相对于只有恒定的比特率和恒定的PSNR编码,它被证明可以获得更好的PSNR/比特率比。我们还展示了该算法在用户在异构网络(WLAN和UMTS)之间漫游的实际场景中的工作情况。同样在这种情况下,该算法被证明可以实现高质量/比特率比。
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引用次数: 1
Neural network based framework for goal event detection in soccer videos 基于神经网络的足球视频目标事件检测框架
Pub Date : 2005-12-12 DOI: 10.1109/ISM.2005.83
Kasun Wickramaratna, Min Chen, Shu‐Ching Chen, M. Shyu
In this paper, a neural network based framework for semantic event detection in soccer videos is proposed. The framework provides a robust solution for soccer goal event detection by combining the strength of multimodal analysis and the ability of neural network ensembles to reduce the generalization error. Due to the rareness of the goal events, the bootstrapped sampling method on the training set is utilized to enhance the recall of goal event detection. Then a group of component networks are trained using all the available training data. The precision of the detection is greatly improved via the following two steps. First, a pre-filtering step is employed on the test set to reduce the noisy and inconsistent data, and then an advanced weighting scheme is proposed to intelligently traverse and combine the component network predictions by taking into consideration the prediction performance of each network. A set of experiments are designed to compare the performance of different bootstrapped sampling schemes, to present the strength of the proposed weighting scheme in event detection, and to demonstrate the effectiveness of our framework for soccer goal event detection.
本文提出了一种基于神经网络的足球视频语义事件检测框架。该框架通过结合多模态分析的强度和神经网络集成的能力来减少泛化误差,为足球进球事件检测提供了一个鲁棒的解决方案。由于目标事件的稀缺性,利用训练集上的自举抽样方法来提高目标事件检测的召回率。然后使用所有可用的训练数据训练一组组件网络。通过以下两个步骤,大大提高了检测的精度。首先对测试集进行预滤波,减少噪声和不一致数据,然后提出一种先进的加权方案,综合考虑各网络的预测性能,对各分量网络预测进行智能遍历和组合。设计了一组实验来比较不同自举采样方案的性能,展示所提出的加权方案在事件检测中的强度,并证明我们的框架在足球目标事件检测中的有效性。
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引用次数: 30
期刊
Seventh IEEE International Symposium on Multimedia (ISM'05)
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