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164
Rate-distortion methods for image and video compression
- IEEE Signal Process. Mag. 1998
"... In this paper we provide an overview of rate-distortion (R-D) based optimization techniques and their practical application to image and video coding. We begin with a short discussion of classical rate-distortion theory and then we show how in many practical coding scenarios, such as in standards-co ..."
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Cited by 224 (7 self)
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In this paper we provide an overview of rate-distortion (R-D) based optimization techniques and their practical application to image and video coding. We begin with a short discussion of classical rate-distortion theory and then we show how in many practical coding scenarios, such as in standards-compliant coding environments, resource allocation can be put in an R-D frame-work. We then introduce two popular techniques for resource allocation, namely, Lagrangian optimization and dynamic programming. After a discussion of these two techniques as well as some of their extensions, we conclude with a quick review of recent literature in these areas citing a number of applications related to image and video compression and transmission. We
Real-Time Internet Video Using Error Resilient Scalable Compression and TCP-Friendly Transport Protocol
, 1999
"... We introduce a point to point real-time video transmission scheme over the Internet combining a low-delay TCP-friendly transport protocol in conjunction with a novel compression method that is error resilient and bandwidth-scalable. Compressed video is packetized into individually decodable packets ..."
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Cited by 179 (9 self)
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We introduce a point to point real-time video transmission scheme over the Internet combining a low-delay TCP-friendly transport protocol in conjunction with a novel compression method that is error resilient and bandwidth-scalable. Compressed video is packetized into individually decodable packets of equal expected visual importance. Consequently, relatively constant video quality can be achieved at the receiver under lossy conditions. Furthermore, the packets can be truncated to instantaneously meet the time varying bandwidth imposed by a TCP-friendly transport protocol. As a result, adaptive flows that are friendly to other Internet traffic are produced. Actual Internet experiments together with simulations are used to evaluate the performance of the compression, transport, and the combined schemes.
The MPEG-4 Fine-Grained Scalable Video Coding Method for Multimedia Streaming over IP
- IEEE Transactions on Multimedia
, 2001
"... Real-time streaming of audiovisual content over the Internet is emerging as an important technology area in multimedia communications. Due to the wide variation of available bandwidth over Internet sessions, there is a need for scalable video coding methods and (corresponding) flexible streaming app ..."
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Cited by 157 (13 self)
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Real-time streaming of audiovisual content over the Internet is emerging as an important technology area in multimedia communications. Due to the wide variation of available bandwidth over Internet sessions, there is a need for scalable video coding methods and (corresponding) flexible streaming approaches that are capable of adapting to changing network conditions in real time. In this paper, we describe a new scalable video-coding framework that has been adopted recently by the MPEG-4 video standard. This new MPEG-4 video approach, which is known as Fine-Granular-Scalability (FGS), consists of a rich set of video coding tools that support quality (i.e., SNR), temporal, and hybrid temporal-SNR scalabilities. Moreover, one of the desired features of the MPEG-4 FGS method is its simplicity and flexibility in supporting unicast and multicast streaming applications over IP.
Scalable compression and transmission of Internet multicast video
, 1996
"... In just a few years the "Internet Multicast Backbone", or MBone, has risen from a small, research curiosity to a large scale and widely used communications infrastructure. A driving force behind this growth was our development of multipoint audio, video, and shared whiteboard conferencing ..."
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Cited by 106 (4 self)
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In just a few years the "Internet Multicast Backbone", or MBone, has risen from a small, research curiosity to a large scale and widely used communications infrastructure. A driving force behind this growth was our development of multipoint audio, video, and shared whiteboard conferencing applications that are now used daily by the large and growing MBone community. Because these real-time media are transmitted at a uniform rate to all the receivers in the network, the source must either run below the bottleneck rate or overload portions of the multicast distribution tree. In this dissertation, we propose a solution to this problem by moving the burden of rate-adaptation from the source to the receivers with a scheme we call Receiver-driven Layered Multicast, or RLM. In RLM, a source distr...
Video Multicast Using Layered FEC and Scalable Compression.
- IEEE Transactions on Circuits and Systems for Video Technology,
, 2001
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VBR video: Trade-offs and potentials
, 1998
"... In this paper, we examine the transport and storage of video compressed with a variable bit rate (VBR). We focus primarily on networked video, although we also briefly consider other applications of VBR video, including satellite transmission (channel sharing), playback of stored video, and wirel ..."
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Cited by 84 (3 self)
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In this paper, we examine the transport and storage of video compressed with a variable bit rate (VBR). We focus primarily on networked video, although we also briefly consider other applications of VBR video, including satellite transmission (channel sharing), playback of stored video, and wireless transport. Packet video research requires careful integration between the network and the video systems; however, a major stumbling block has resulted because commonly used terms are often interpreted differently by the video and networking communities.
Experiments with a Layered Transmission Scheme over the Internet
, 1997
"... Combining hierarchical coding of data with receiver-driven control appears to be an attractive scheme for the multicast transmission of audio/video flows in a heterogeneous multicast environment such as the Internet. However, little experimental data is available regarding the actual performance of ..."
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Cited by 82 (5 self)
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Combining hierarchical coding of data with receiver-driven control appears to be an attractive scheme for the multicast transmission of audio/video flows in a heterogeneous multicast environment such as the Internet. However, little experimental data is available regarding the actual performance of such schemes over the Internet. Previous work such as that on receiver driven layered multicast uses join experiments to choose the best quality signal a receiver can subscribe to. In this paper, we present a receiver-based multicast rate control mechanism based on a recently proposed TCP-friendly unicast mechanism. We have implemented this mechanism and evaluate its performance in conjunction with a simple layered audio coding scheme. We find that it has interesting convergence and performance properties, but also bring out its limitations.
Toward a Common Infrastructure for Multimedia-Networking Middleware
, 1997
"... Real-time multimedia streams like audio and video are now integral data types in modern programming environments. Although a great deal of research has investigated effective and efficient programming support for manipulating such streams and although the design of digital media "middleware&quo ..."
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Cited by 81 (13 self)
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Real-time multimedia streams like audio and video are now integral data types in modern programming environments. Although a great deal of research has investigated effective and efficient programming support for manipulating such streams and although the design of digital media "middleware" is fairly well understood, no widely available or commonly accepted programming model exists within the research community. We believe this lack of common practice impedes our collective progress because it prevents disparate research groups from easily leveraging each other's work. In this paper, we propose a solution to this problem that combines the best features of a number of existing multimedia toolkits --- Berkeley's Continuous Media Toolkit, MIT's VuSystem, and the LBL/UCB MBone tools --- into a fine-grained, extensible, and highperformance toolkit. We describe the convergence of these three toolkits into a common programming infrastructure and argue that the availability and acceptance of ...
Survey of error recovery techniques for IP-based audio-visual multicast applications
- IEEE Network
, 1997
"... IP-based audio-visual multicast applications are gaining increasing interest since they can be realised using inexpensive network services that offer no guarantees for loss or delay. When using network services that do not guarantee the Quality of Service (QoS) required by audio-visual applications, ..."
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Cited by 75 (2 self)
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IP-based audio-visual multicast applications are gaining increasing interest since they can be realised using inexpensive network services that offer no guarantees for loss or delay. When using network services that do not guarantee the Quality of Service (QoS) required by audio-visual applications, the recovery from losses due to congestion in the network is a key problem that must be solved. This survey gives an overview of existing transport-layer error control mechanisms and discusses their suitability for use in IP-based networks. Additionally, the impact of IP over ATM on the requirements of error control mechanisms is discussed. Different network scenarios are used to assess the performance of retransmission-based error correction and forward error correction. 1
Error Control Techniques for Interactive Low-bit Rate Video Transmission over the Internet
- ACM SIGCOMM
, 1998
"... A new retransmission-based error control technique is presented that does not incur any additional latency in frame playout times, and hence are suitable for interactive applications. It takes advantage of the motion prediction loop employed in most motion compensationbased codecs. By correcting err ..."
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Cited by 75 (4 self)
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A new retransmission-based error control technique is presented that does not incur any additional latency in frame playout times, and hence are suitable for interactive applications. It takes advantage of the motion prediction loop employed in most motion compensationbased codecs. By correcting errors in a reference frame caused by earlier packet loss, it prevents error propagation. The technique rearranges the temporal dependency of frames so that a displayed frame is referenced for the decoding of its succeeding dependent frames much later than its display time. Thus, the delay in repairing lost packets can be effectively masked out. The developed technique is combined with layered video coding to maintain consistently good video quality even under heavy packet loss. Through the results of extensive Internet experiments, the paper shows that layered coding can be very effective when combined with the retransmissionbased error control technique for low-bit rate transmission over best...
