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Application-layer mobility support for streaming real-time media

Application-layer mobility support for streaming real-time m...

引用

IEEE Conference on Wireless Communications and Networking

作者： J. Kristiansson P. Parnes Department of Computer Science & Electrical Engineering Media Technology Luleå University of Technology Lulea Sweden

This paper presents an UDP-based socket extension called the resilient mobile socket (RMS), which provides application-layer mobility support by encapsulating other sockets into a new aggregated socket abstraction. Encapsulated sockets can then be added or removed without disturbing running applications. RMS also provides a method for soft handovers where several encapsulated sockets are used simultaneously during a handover. As a proof of concept, a working prototype has been built by integrating RMS with Marratech Pro, a commercially available e-meeting application. This prototype has been used to evaluate RMS and to investigate how GSM audio quality is affected by handovers. The result from the investigation shows that soft handovers can be executed without loosing packets or causing extra latency, while a hard handover in average took around 200 ms to complete. This indicates that proactive handovers and redundancy are important, but that more work must he done to predict disconnections.

关键词： Streaming media Sockets Prototypes Delay computer science Mobile computing GSM Land mobile radio cellular systems Internet telephony Mobile communication

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Eventcasting with a wearable computer

Eventcasting with a wearable computer

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International Conference on Distributed Computing Systems Workshop

作者： M. Drugge M. Nilsson K. Synnes P. Parnes Department of Computer Science & Electrical Engineering Division of Media Technology Luleå University of Technology Lulea Sweden

Traditionally, interaction methods for wearable computers have focused on input to the computer itself, yet little has been done when it comes to allowing interaction with the surrounding environment. Pervasive computing, on the other hand, offers access to computational power from any place all the time, yet most interaction techniques utilize either physical hardware or monitoring of the user in order to receive input. This paper presents a novel form of interaction by which a wearable computer user can interact with and control a pervasive computing environment in a natural and intuitive manner. Using sensors, the user can be allowed to literally "throw" events into the environment as a way of interacting with devices and computers.

关键词： Wearable computers Pervasive computing Pharmaceutical technology Hardware Wearable sensors Humans computer science Power engineering computing Physics computing computerized monitoring

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Methods for interrupting a wearable computer user

Methods for interrupting a wearable computer user

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International Symposium on Wearable computers

作者： M. Drugge M. Nilsson U. Liljedahl K. Synnes P. Parnes Department of Computer Science & Electrical Engineering Division of Media Technology Luleå University of Technology Lulea Sweden Department of Computer Science & Electrical Engineering Division of Computer Science and Networking Luleå University of Technology Lulea Sweden

A wearable computer equipped with a head-mounted display allows its user to receive notifications and advice that is readily visible in her field of view. While needless interruption of the user should be avoided, there are times when the information is of such importance that it must demand the user's attention. As the user is mobile and likely interacts with the real world when these situations occur, it is important to know in what way the user can be notified without increasing her cognitive workload more than necessary. To investigate ways of presenting information without increasing the cognitive workload of the recipient, an experiment was performed testing different approaches. The experiment described in this paper is based on an existing study of interruption of people in human-computer interaction, but our focus is instead on finding out how this applies to wearable computer users engaged in real world tasks.

关键词： Wearable computers Pharmaceutical technology computer science Pervasive computing computer displays Performance evaluation Testing computer networks Fires Interference

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Adaptive source and channel re-encoding to maximize tandem link performance

Adaptive source and channel re-encoding to maximize tandem l...

引用

International Symposium on Communications Control and Signal Processing (ISCCSP)

作者： J.D. Gibson Bo Wei Media Arts Technology Program and Electrical Engineering University of California Santa Barbara CA USA Department of Electrical Engineering Southern Methodist University Dallas TX USA

We investigate the issues in voice communications over tandem heterogeneous network connections, which require multiple source decoding/encoding operations. While it is known from speech coding research that tandem connections of different speech coders can extract a substantial performance penalty, it is shown that the use of error concealment and source re-encoding can improve system performance in the presence of channel impairments such as bit errors and packet losses. A new protocol is proposed for wireless LANs that reduces retransmissions and discarded packets, and hence reduces system delay and facilitates packet loss concealment.

关键词： Decoding Speech coding Speech codecs Internet telephony Bit rate Nonhomogeneous media Art computer networks Performance loss System performance

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5th International Symposium on Music Information Retrieval, ISMIR 2004

作者： Cruz, Fernando W. Ferneda, Edilson Brandão, Márcio Costa, Evandro De B. De Almeida, Hyggo O. Da Cunha, Murilo B. De Sousa, Rafael T. Denicol, João Ricardo E. Da Silva, Carlos Alan P. Catholic University of Brasília Computer Science Department Brasília Brazil University of Brasília Computer Science Department Brasília Brazil Federal University of Alagoas Information Technology Department Maceió Brazil Federal University of Campina Grande Electrical Engineering Department Campina Grande Brazil University of Brasília Information Science Department Brasília Brazil University of Brasília Electrical Engineering Department Brasília Brazil Federal University of Campina Grande Arts Department Campina Grande Brazil

This poster presents a digital library proposal conceived for people interested in acquiring knowledge about Brazilian popular music harmony, particularly in Choro. This Brazilian musical style is a complex popular music form based on improvisation, although it contains classical music elements such as the counterpoint. We are proposing two ways of accessing the music virtual library content: a guided navigation mode, in which users interact with a cooperative Web-based learning system;and a free navigation mode, in which users can make their own queries, both through browsers or client applications. © 2004 Universitat Pompeu Fabra.

关键词： Music

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Joint semiblind channel estimation and data detection for MIMO-OFDM systems

Joint semiblind channel estimation and data detection for MI...

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Proceedings of the IEEE Circuits and Systems Symposium on Emerging Technologies

作者： Jiang Yue Kyeong Jin Kim T. Reid J.D. Gibson Department of Electrical Engineering Southern Methodist University Dallas TX USA Nokia Research Center TX Nokia Research Center Irving TX USA Media Arts and Technology Program and Electrical & Computer Engineering University of California Santa Barbara Santa Barbara CA

In MIMO systems, where multiple antennas are used at both transmitter and receiver to achieve high spectral efficiency, channel impulse responses are often assumed to be constant over a block or packet. This assumption of block stationarity on channels is valid for most fixed wireless scenarios. However, for communications in a high mobility environment, the assumption will result in considerable performance degradation. In this paper, we focus on channel estimation for a MIMO system with OFDM transmission technique. In our system, pilots are placed on N/sub P/ subcarriers for a novel channel estimation at the receiver with Kalman filters. With the channels estimated by a Kalman filter, we apply the QRD-M algorithm for suboptimal data detection with reasonable computational cost. The outputs of data detection are fed back to another Kalman filter for an improved channel estimation. By alternatively and iteratively using these two Kalman filters, a better channel estimation can be obtained. A simulation result of the bit error rate shows the improvement of 10 dB in SNR with four iterations over noniteration.

关键词： Channel estimation MIMO Receiving antennas Transmitting antennas Transmitters Degradation OFDM Iterative algorithms Computational efficiency Computational modeling

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Vehicle detection and tracking using acoustic and video sensors

Vehicle detection and tracking using acoustic and video sens...

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International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： R. Chellappa Gang Qian Qinfen Zheng Center for Automation Research Institute for Advanced Computer Studies University of Maryland College Park MD USA Department of Electrical Engineering and Arts Media and Engineering Program Arizona State University Tempe AZ USA

Multimodal sensing has attracted much attention in solving a wide range of problems, including target detection, tracking, classification, activity understanding, speech recognition, etc. In surveillance applications, different types of sensors, such as video and acoustic sensors, provide distinct observations of ongoing activities. We present a fusion framework using both video and acoustic sensors for vehicle detection and tracking. In the detection phase, a rough estimate of target direction-of-arrival (DOA) is first obtained using acoustic data through beam-forming techniques. This initial DOA estimate designates the approximate target location in video. Given the initial target position, the DOA is refined by moving target detection using the video data. Markov chain Monte Carlo techniques are then used for joint audio-visual tracking. A novel fusion approach has been proposed for tracking, based on different characteristics of audio and visual trackers. Experimental results using both synthetic and real data are presented. Improved tracking performance has been observed by fusing the empirical posterior probability density functions obtained using both types of sensors.

关键词： Vehicle detection Acoustic sensors Target tracking Direction of arrival estimation Object detection Multimodal sensors Speech recognition Surveillance Acoustic signal detection Phase detection

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An adaptive DSP processor for high-efficiency computing MPEG-4 video encoder

An adaptive DSP processor for high-efficiency computing MPEG...

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IEEE International Symposium on Circuits and Systems (ISCAS)

作者： Li-Hsun Chen O.T.-C. Chen Teng-Yi Wang Chi-Lung Wang Signal and Media Laboratories Department of Electrical Engineering National Chung Cheng University Chiayi Taiwan Computer System Development Department Computer & Communications Research Laboratories Industrial Technology and Research Institute Hsinchu Taiwan

An adaptive digital signal processor (DSP) is proposed to realize the MPEG-4 video encoder at a high ratio of computation power versus hardware cost. First, software analyses are performed on the MPEG-4 video encoder, looking at the function and instruction levels. Analytical results from the function-level analysis show that the motion estimation inside the MPEG-4 video encoder has high computational complexity. As for the instruction-level analysis on the MPEG-4 video encoder without motion estimation, results reveal that the DSP equipped with 5 arithmetic logic units (ALUs), 1 multiplier and 2 Load/Store units has a higher computation performance than the other architectures with 8 functional units. Furthermore, the design of the adaptive mechanism can be incorporated in the said functional units. The suggestion is to group 4 ALUs and a multiplier into a special functional unit to exclusively process motion estimation. With the fact that a multiplier can be constructed by using multiple adders, the adaptable structure is also designed so that the numbers of adders and multipliers required can be dynamically changed to increase parallelism capability of local instructions, especially in discrete cosine transform and inverse discrete cosine transform. In comparing to conventional DSPs, the proposed adaptive DSP shows the best ratio of computation power over hardware cost to realize the MPEG-4 video encoder.

关键词： Digital signal processing MPEG 4 Standard Motion estimation Hardware Costs Performance analysis Discrete cosine transforms Digital signal processors Software performance Computational complexity

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A low-complexity architecture of inverse fast Fourier transform for XDS

A low-complexity architecture of inverse fast Fourier transf...

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Midwest Symposium on Circuits and Systems (MWSCAS)

作者： Hsin-Horng Chen O.T.-C. Chen Heng-Cheng Yeh Cheng-Shing Wu Signal and Media Laboratories Department of Electrical Engineering National Chung Cheng University Chiayi Taiwan Transmission System Department Computer & Communications Research Laboratories Industrial Technology and Research Institute Hsinchu Taiwan

In this work, a low-complexity architecture of the inverse fast Fourier transform (IFFT) using the serial-input-parallel-output data flow is designed for digital subscriber line for any class (XDSL). Since input data in the XDSL are symmetric, the computation of the IFFT can be reduced to multiplication accumulation operations in the real part. Furthermore, consideration is taken to process all multiplication operations together to lower hardware complexity. By separating all multiplication and accumulation operations into two groups, only an interface is needed to be the bridge between the two. The coefficients of the IFFT are represented by the canonic signed digits (CSDs) so that their shared terms are derived to reduce the number of adders for accomplishing multiplication operations. By using multiplexors to do the interface, the computation results from the multiplication part are distributed to their corresponding accumulators according to a time sequence. In addition, the size of the word length can be adequately selected to match the required signal-to-noise ratio (SNR). As compared to the conventional IFFT architectures, the proposed IFFT architecture can have the least hardware complexity at the same throughput rate and SNR performance.

关键词： Fast Fourier transforms Hardware Signal processing Concurrent computing computer architecture Discrete Fourier transforms Discrete cosine transforms Bridges Throughput OFDM modulation

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Comparing objective feature statistics of speech for classifying clinical depression

Comparing objective feature statistics of speech for classif...

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Annual International Conference of the IEEE engineering in Medicine and Biology Society (EMBC)

作者： E.I.I. Moore M. Clements J. Peifer L. Weisser School of Electrical and Computer Engineering Georgia Institute of Technology Atlanta GA USA Interactive Media Technology Center Georgia Institute of Technology Atlanta GA USA Department of Psychiatry and Behavioral Health Medical College of Georgia Atlanta GA USA

Human communication is saturated with emotional context that aids in interpreting a speakers mental state. Speech analysis research involving the classification of emotional states has been studied primarily with prosodic (e.g., pitch, energy, speaking rate) and/or spectral (e.g., formants) features. Glottal waveform features, while receiving less attention (due primarily to the difficulty of feature extraction), have also shown strong clustering potential of various emotional and stress states. This study provides a comparison of the major categories of speech analysis in the application of identifying and clustering feature statistics from a control group and a patient group suffering from a clinical diagnosis of depression.

关键词： Statistics Speech analysis Humans Stress Medical services Psychology Speech processing Biomedical engineering Psychiatry Medical diagnostic imaging

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