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Circuits and Systems for Video Technology, IEEE Transactions on

Issue 9 • Date Sep 2001

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Displaying Results 1 - 8 of 8
  • Multiple description coding of motion fields for robust video transmission

    Publication Year: 2001 , Page(s): 999 - 1010
    Cited by:  Papers (21)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (304 KB) |  | HTML iconHTML  

    In many video-coding standards, the motion vector field is one of the most important data in the compressed bitstream, and its loss can lead to severe degradation in the decoded picture quality. We propose the multiple description motion coding (MDMC) algorithm to enhance the robustness of the motion vector field against transmission errors. In MDMC, the motion vector field is encoded into two descriptions, which are transmitted over distinct channels to the decoder. The decoder is designed to provide an acceptable quality prediction image, even if one of the descriptions is lost during the transmission. Moreover, the decoder can reconstruct a higher quality prediction image, when both the descriptions are received without error. A complete multiple description video coder, based on the MDMC, is implemented by modifying the syntax of the H.263 standard, and tested intensively in a realistic error-prone environment. It is shown that the proposed algorithm provides much better objective and subjective performances than the H.263 coder in the error-prone environment View full abstract»

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  • Robust transmission of video sequence using double-vector motion compensation

    Publication Year: 2001 , Page(s): 1011 - 1021
    Cited by:  Papers (13)  |  Patents (8)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (280 KB) |  | HTML iconHTML  

    This paper proposes a motion compensation (MC) algorithm for robust transmission of video sequence, called the double-vector motion compensation (DMC). In the DMC, each block B in a frame is predicted from the weighted superposition of two blocks in the previous two frames, using two motion vectors. Therefore, when one of these two blocks is corrupted during the transmission, the decoder can efficiently suppress its error propagation to the subsequent frames, by predicting B only from the other block. It is shown by analysis that the DMC algorithm yields significantly lower error bounds for the subsequent frames than the conventional MC technique. Furthermore, the DMC algorithm can be combined with an effective concealment algorithm, which is capable of recovering very severe transmission errors, such as loss of an entire frame. A complete video coder, based on the DMC, is implemented by modifying the MC syntax of the H.263 standard and tested intensively in a realistic error prone environment. It is shown that the proposed algorithm provides much better objective and subjective performances than the H.263 coder in the error-prone environment View full abstract»

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  • A resolution and frame-rate scalable subband/wavelet video coder

    Publication Year: 2001 , Page(s): 1035 - 1044
    Cited by:  Papers (13)  |  Patents (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (184 KB) |  | HTML iconHTML  

    A source coding algorithm is presented which delivers a spatiotemporal scalable encoded bitstream. Motion compensated temporal filtering is combined with a spatial subband/wavelet pyramid to provide an efficient 3-D multiresolution representation. Error-feedback hierarchical coding permits near-optimal results for each subvideo. The proposed ordering of refinement addresses the noncommutative property of motion compensated temporal filtering and spatial subband/wavelet analysis. Finally, adaptive conditional arithmetic coding of quantizer significance maps increases coding efficiency. Experimental results demonstrate a significant improvement in performance over earlier published algorithms. The code-and-refine nature of our hierarchical algorithm makes it possible to use the algorithm as a scalable extension to other source coding algorithms. Finally, the complexity of the algorithm is modest and is well suited to parallel implementation View full abstract»

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  • 3-D wavelet coding of video with arbitrary regions of support

    Publication Year: 2001 , Page(s): 1063 - 1068
    Cited by:  Papers (33)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (208 KB) |  | HTML iconHTML  

    We examine 3-D wavelet coding of video with arbitrary regions of support (AROS). A critically sampled wavelet transform is applied to the AROS and a modified 3-D set partitioning in hierarchical trees (SPIHT) algorithm is used to quantize and code the wavelet coefficients in the AROS only. Experiments show that, for typical MPEG-4 pre-segmented sequences, our proposed method can achieve a gain of up to 5.6 dB in average PSNR at the same rate over 3-D SPIHT coding of regular volumes that embed the AROS of the given video sequences View full abstract»

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  • Rate control for VBR video over ATM: simplification and implementation

    Publication Year: 2001 , Page(s): 1045 - 1058
    Cited by:  Papers (8)  |  Patents (16)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (296 KB) |  | HTML iconHTML  

    Rate control is essential to ensure high-quality video transmission over asynchronous transfer mode (ATM) networks; both source and channel rates must not violate buffer or network traffic constraints in order to guarantee decoded video quality. To date, source and channel rate selection has been performed jointly because the rates are related through buffer and network constraints and therefore appear to be interdependent. It is shown that selection of these rates can be separated and the rate control problem is therefore simplified, allowing sequential source and channel rate selection without the need for iteration. A rate control algorithm is proposed which implements noniterative, separate sequential selection of source and channel rates in order to minimize the distortion variation between frames, subject to all buffer and network constraints. In particular, the leaky bucket is used as the traffic policing mechanism. Achieving a target rate for individual frames is facilitated by using a rate-distortion optimized video coder which allows coding to an exactly specified rate or distortion. Channel rate selection is performed following source rate selection with the goal of providing smooth traffic to the network, thereby allowing a high statistical multiplexing gain. Experiments demonstrate that the proposed noniterative rate control algorithm, when implemented using the optimized coder, achieves consistent video quality while providing smooth traffic to the network View full abstract»

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  • AMISP: a complete content-based MPEG-2 error-resilient scheme

    Publication Year: 2001 , Page(s): 989 - 998
    Cited by:  Papers (20)  |  Patents (5)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (208 KB) |  | HTML iconHTML  

    We address a new error-resilient scheme for broadcast quality MPEG-2 video streams to be transmitted over lossy packet networks. A new scene-complexity adaptive mechanism, namely Adaptive MPEG-2 Information Structuring (AMIS) is introduced. AMIS modulates the number of resynchronization points (i.e., slice headers and intra-coded macroblocks) in order to maximize the perceived video quality, assuming that the encoder is aware of the underlying packetization scheme, the packet loss probability (PLR), and the error-concealment technique implemented at the decoding side. The end-to-end video quality depends both on the encoding quality and the degradation due to data loss. Therefore, AMIS constantly determines the best compromise between the rate allocated to encode pure video information and the rate aiming at reducing the sensitivity to packet loss. Experimental results show that AMIS dramatically outperforms existing structuring techniques, thanks to its efficient adaptivity. We then extend AMIS with a forward-error-correction (FEC)-based protection algorithm to become AMISP. AMISP triggers the insertion of FEC packets in the MPEG-2 video packet stream. Finally, the performances of the AMISP scheme in an MPEG-2 over RTP/UDP/IP scenario are evaluated View full abstract»

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  • Classified zerotree wavelet image coding and adaptive packetization for low-bit-rate transport

    Publication Year: 2001 , Page(s): 1022 - 1034
    Cited by:  Papers (13)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (312 KB) |  | HTML iconHTML  

    A novel robust image coding and adaptive packetization algorithm suitable for very low-bit-rate transport is suggested. This algorithm can be applied to any zerotree-based encoder, such as the embedded zerotree wavelet coder of Shapiro (1993) and set partitioning in hierarchical trees by Said and Pearlman (1996). A very explicit segmentation and packetization method of an image bitstream, where the lowest frequency subband is separately encoded from the higher frequency subbands for unequal protection over a noisy channel, is proposed. The trees in the higher frequency subbands are split, classified, and assembled for efficient image coding and packetization according to their initial threshold and subband. The use of these classified trees enables one to make robust packets, while giving priority to some packets. In practice, each packet has a different initial threshold and can be decoded independently. In spite of additional overhead bits required for packetization, the algorithm reported is comparable to the original zerotree-based image coders at low bit rates. Additionally, simulation results show that the new method is resilient under severe packet losses View full abstract»

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  • Impact of audio on subjective assessment of video quality in videoconferencing applications

    Publication Year: 2001 , Page(s): 1059 - 1062
    Cited by:  Papers (6)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (94 KB) |  | HTML iconHTML  

    In the real world, we commonly receive information simultaneously through two or more senses, with the brain fusing this data to produce a single coherent message. Lip-reading is one example of this phenomenon. Laboratory studies, on the other hand, often measure the response to a stimulus by a single sense and extrapolate these results to predict real-world behavior. In this paper, we show that semantics have a significant impact on viewers' sensitivity to the quality of a video sequence for spatially separated parts of the sequence and, more importantly, that this difference in sensitivity can be changed by the presence of an audio signal. This result is important for any testing of subjects' responses to visual material. One example is the subjective assessment of the quality of video in an audio-visual communications system (such as television or videoconferencing) View full abstract»

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Aims & Scope

The emphasis is focused on, but not limited to:
1. Video A/D and D/ A
2. Video Compression Techniques and Signal Processing
3. Multi-Dimensional Filters and Transforms
4. High Speed Real-Tune Circuits
5. Multi-Processors Systems—Hardware and Software
6. VLSI Architecture and Implementation for Video Technology 

 

Full Aims & Scope

Meet Our Editors

Editor-in-Chief
Dan Schonfeld
Multimedia Communications Laboratory
ECE Dept. (M/C 154)
University of Illinois at Chicago (UIC)
Chicago, IL 60607-7053
tcsvt-eic@tcad.polito.it

Managing Editor
Jaqueline Zelkowitz
tcsvt@tcad.polito.it