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Selected Topics in Signal Processing, IEEE Journal of

Issue 2 • Date Aug. 2007

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Displaying Results 1 - 17 of 17
  • Table of contents

    Page(s): C1
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  • IEEE Journal of Selected Topics in Signal Processing publication information

    Page(s): C2
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  • Introduction to the Issue on Network-Aware Multimedia Processing and Communications

    Page(s): 217 - 219
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  • Distributed Source Coding for Multimedia Multicast Over Heterogeneous Networks

    Page(s): 220 - 230
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (967 KB) |  | HTML iconHTML  

    Real-time multimedia multicast over wireless networks is an exciting application that has generated a lot of interest recently. Its main challenge lies in the stringent bandwidth and time-delay requirements of real-time multimedia and severe impairments of the wireless channels. We develop a network-aware cross-layer design for multimedia multicast over heterogeneous wireless-wireline networks, that leverages the knowledge on network information theory, multimedia processing, error control, and networking. In particular, the encoded multimedia data are broadcast to multiple Internet servers over a wireless radio link. Each server merely compresses the signal it has received using distributed source coding by exploiting mutual correlation among signals received at different servers. The receiver collects bitstreams from the servers before performing joint decoding. We provide an algorithm for optimal nonuniform scalar quantizer design at the server side that minimizes the required rate under the decoder bit error rate constraint. For scalable multimedia codes, we develop a joint source-channel coding scheme which combines error-protection at the base station and distributed source coding at the servers. Our experimental results show significant performance improvements over conventional solutions due to spatial diversity and distributed source coding gains. View full abstract»

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  • A Flexible Multiple Description Coding Framework for Adaptive Peer-to-Peer Video Streaming

    Page(s): 231 - 245
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1568 KB) |  | HTML iconHTML  

    Efficient peer-to-peer (P2P) video streaming is a challenging task due to time-varying nature of both the number of available peers and network/channel conditions. This paper proposes a novel adaptive P2P video streaming system, which features: (i) a new flexible multiple-description coding (F-MDC) framework, such that the number of base and enhancement descriptions, and the rate and redundancy level of each description can be adapted on the fly (by only post-processing of the encoded bitstream), and (ii) a new adaptive TCP-friendly rate-controlled (TFRC), on-demand, many-to-one P2P video streaming solution based on the proposed F-MDC framework. We extend the highly scalable video coder , to MDC within the proposed F-MDC framework. Optimization of the design parameters of the proposed F-MDC method is discussed within the context of the proposed adaptive P2P streaming solution, where the number and quality of available streaming peers/paths are a priori unknown and vary in time. Experimental results, by means of NS-2 network simulation of a P2P video streaming system, show that adaptation of the number, type of descriptions and the rate and redundancy level of each description according to network conditions yields significantly superior performance when compared to other scalable MDC schemes using a fixed number of descriptions/layers with fixed rate and redundancy level. View full abstract»

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  • Congestion Control for Scalable Video Streaming Using the Scalability Extension of H.264/AVC

    Page(s): 246 - 253
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (915 KB) |  | HTML iconHTML  

    This paper presents a streaming system using the scalability extension of H.264/AVC. The system provides a congestion control algorithm, which is supported by a channel bandwidth estimation running on the client. It uses retransmission only for packets from the base layer to limit the burden on the congested network. The bandwidth estimation allows the transmission rate to be quickly adjusted to the currently available bandwidth of the network. Compared to binomial congestion control, the proposed system allows for shorter start-up times and faster data rate adaptation. The paper describes the components of this streaming system and the results of experiments with competing UDP and TCP applications showing that the proposed approach reaches a throughput at least 50% higher than existing congestion control algorithms for streaming video without using more than the fair share of the bandwidth of the bottleneck link. View full abstract»

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  • TeleMorph: A Fuzzy Logic Approach to Network-Aware Transmoding in Mobile Intelligent Multimedia Presentation Systems

    Page(s): 254 - 263
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (723 KB) |  | HTML iconHTML  

    Mobile intelligent multimedia presentation systems are limited by various constraints including mobile network characteristics, mobile device capabilities, and user preferences. Those presentation systems which incorporate remotely stored multimedia rely on the bandwidth which is available during actual content transmission on the connecting mobile network. One approach to deal with this is to transcode content, thus reducing its data rate requirement, although this technique is inherently limited by the lowest acceptable quality of that media element. Alternatively, content can be transmoded to different modalities with lower bandwidth requirements. TeleMorph, a cross-modality adaptation control platform is detailed in this paper. The main premise of TeleMorph is that cross-modality adaptation decisions in mobile presentation systems must occur with primary consideration for bandwidth fluctuations. TeleMorph has been implemented as a network-aware fuzzy inference system that controls cross-modality adaptations between multimodal audio-video, audio-images, and images-text presentations, as well as unimodal audio and text presentations. Initially, a brief introduction to Intelligent Multimedia and Mobile Intelligent Multimedia is given, and related systems discussed. TeleTuras, a tourist information application implemented as a testbed for TeleMorph, is utilized for objective and subjective evaluations through its integration of six disparate test scenarios that incorporate various media types and qualities. Positive results for each evaluation technique, based on specific metrics, are also presented. In addition, future work on TeleMorph is also detailed. View full abstract»

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  • A Pricing Mechanism for Resource Allocation in Wireless Multimedia Applications

    Page(s): 264 - 279
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (658 KB) |  | HTML iconHTML  

    We consider the problem of multiuser resource allocation for wireless multimedia applications deployed by autonomous and noncollaborative wireless stations (WSTAs). Existing resource allocation solutions for WLANs are not network-aware and do not take into account the selfish behavior of individual WSTAs. Specifically, the selfish WSTAs can manipulate the network by untruthfully representing their private information (i.e., video characteristics, experienced channel conditions, and deployed streaming strategies). This often results in inefficient resource allocations. To overcome this obstacle, we present a pricing mechanism for message exchanges between the WSTAs and the Central Spectrum Moderator (CSM). The messages represent network-aware resource demands and corresponding prices. We prove that the message exchanges reach the Nash equilibrium and that the resulting equilibrium messages generate allocations which are efficient, budget balanced, and satisfy voluntary participation. The simulation results verify that these properties hold when the WSTAs behave strategically. Additionally, we evaluate the impact of initial prices and network congestion level on the convergence rate of message exchanges. View full abstract»

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  • Joint Uplink and Downlink Optimization for Real-Time Multiuser Video Streaming Over WLANs

    Page(s): 280 - 294
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1184 KB) |  | HTML iconHTML  

    In this paper, a network-aware and source-aware video streaming system is proposed to support interactive multiuser communications within single-cell and multicell IEEE 802.11 networks. Unlike the traditional streaming video services, the strict delay constraints of an interactive video streaming system pose more challenges. These challenges include the heterogeneity of uplink and downlink channel conditions experienced by different users, the multiuser resource allocation of limited radio resources, the incorporation of the cross-layer design, and the diversity of content complexities exhibited by different video sequences. With the awareness of video content and network resources, the proposed system integrates cross-layer error protection mechanism and performs dynamic resource allocation across multiple users. We formulate the proposed system as to minimize the maximal end-to-end expected distortion received by all users, subject to maximal transmission power and delay constraints. To reduce the high dimensionality of the search space, fast multiuser algorithms are proposed to find the near-optimal solutions. Compared to the strategy without dynamically and jointly allocating bandwidth resource for uplinks and downlinks, the proposed framework outperforms by 2.18~7.95 dB in terms of the average received PSNR of all users and by 3.82~11.50 dB in terms of the lowest received PSNR among all users. Furthermore, the proposed scheme can provide more uniform video quality for all users and lower quality fluctuation for each received video sequence. View full abstract»

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  • Cooperative Source and Channel Coding for Wireless Multimedia Communications

    Page(s): 295 - 307
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (651 KB) |  | HTML iconHTML  

    Past work on cooperative communications has indicated substantial improvements in channel reliability through cooperative transmission strategies. To exploit cooperation benefits for multimedia transmission over slow fading channels, we propose to jointly allocate bits among source coding, channel coding and cooperation to minimize the expected distortion of the reconstructed signal at the receiver. Recognizing that, not all source bits are equally important in terms of the end-to-end distortion, we further propose to protect the more important bits through user cooperation. We compare four modes of transmission that differ in their compression and error protection strategies (single layer or multiple layer source coding with unequal error protection, with versus without cooperation). Our study includes an i.i.d. Gaussian source as well as a video source employing an H.263+ codec. We present an information theoretic analysis for the Gaussian source to investigate the effects of the modulation scheme, bandwidth ratio (number of channel uses per source sample), and average link signal-to-noise ratios on the end-to-end distortion of the four modes studied. The information theoretic observations are validated using practical channel coding simulations. Our study for video considers error propagation in decoded video due to temporal prediction and jointly optimizes a source coding parameter that controls error propagation, in addition to bits for source coding, channel coding and cooperation. The results show that cooperation can significantly reduce the expected end-to-end distortion for both types of source and that layered cooperation provides further improvements and extends the benefits to a wider range of channel qualities. View full abstract»

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  • Predictive Path Switching Control for Improving the Quality of Service in Real-Time Applications

    Page(s): 308 - 318
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (761 KB) |  | HTML iconHTML  

    A congestion avoidance strategy based on the concept of predictive path switching control (PPSC) is proposed for improving the quality-of-service (QoS) in real-time applications, and in particular in Voice over Internet Protocol (VoIP) applications deployed over the public Internet. The proposed PPSC strategy requires the dynamic prediction of traffic congestion levels over all available network paths. This predictive information is then used as an input for a control strategy to select the best available path at each pre-determined time. A study using emulated network paths explores the impact of overall path loss rate and traffic delay signal frequency content on the proposed PPSC strategy. The study reveals that PPSC provides the best QoS improvement for real-time applications if the average over-all loss rate of the available paths is between 5% and 15%, and if the traffic delay signal contains mostly low frequencies. Linear and nonlinear dynamic predictors are developed and used along with actual network data collected from PlanetLab for demonstrating the effectiveness of the PPSC strategy. The results show that PPSC is better than no path switching, with no one dynamic predictor providing the best performance for all case studies. A voting-based control strategy is proposed to overcome this problem. The results demonstrate that the voting-based control results in universally better performance for all cases studied. VoIP packets encoded with Speex, a publicly available encoder, demonstrate that any PPSC strategy is far more effective than no path switching. The proposed voting-based PPSC is moderately more effective than PPSC based on a simple predictor, both resulting in voice quality mostly over 3.0 in MOS, if the available network paths have overall loss rate in the range of 5% and 15%. Though encouraging, the PPSC strategy raises a number of implementation issues, which must still be addressed. In particular, scaling of the PPSC to a large number of networ- k paths will require probing packets that generate significant overhead traffic. These and other issues require further research. View full abstract»

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  • Smart Media Striping Over Multiple Burst-Loss Channels

    Page(s): 319 - 333
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1114 KB) |  | HTML iconHTML  

    We consider a community of multi-homed wireless devices, where each device has both a wireless wide area network (WWAN) interface to connect to the Internet and a wireless local area network (WLAN) interface to connect to its neighbors. Suppose users in the community are interested in receiving the same piece of delay-sensitive media content, and are willing to share their network resources. It is obvious that the community can benefit from the bundling of WWAN links and achieve higher aggregate bandwidth that is not possible with a single user with a single WWAN connection. What is not obvious is that by inverse multiplexing or striping packets across multiple WWAN channels, one can also improve the goodput of delay-sensitive media traffic by striping FEC and ARQ packets across available channels. In this paper, we analyze the potential benefits of striping media traffic and develop algorithms that take advantage of these benefits to optimize the delivery of delay-sensitive media streams to a wireless multi-homed device community. Results show dramatic improvement over naive striping schemes such as weighted round robin both in terms of packet loss ratio, and in terms of peak signal-to-noise ratio for H.264 video streaming. View full abstract»

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  • IEEE Journal of Selected Topics in Signal Processing Information for authors

    Page(s): 334
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  • Special issue on Genomic and Proteomic Signal Processing

    Page(s): 335
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  • Special issue on Distributed Processing in Vision Netwroks

    Page(s): 336
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  • IEEE Signal Processing Society Information

    Page(s): C3
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  • Blank page [back cover]

    Page(s): C4
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Aims & Scope

The Journal of Selected Topics in Signal Processing (J-STSP) solicits special issues on topics that cover the entire scope of the IEEE Signal Processing Society including the theory and application of filtering, coding, transmitting, estimating, detecting, analyzing, recognizing, synthesizing, recording, and reproducing signals by digital or analog devices or techniques.

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Meet Our Editors

Editor-in-Chief
Fernando Pereira
Instituto Superior Técnico