Chi-Yuan Hsu

Bio: Chi-Yuan Hsu is an academic researcher from University of Southern California. The author has contributed to research in topics: Communication channel & Video quality. The author has an hindex of 7, co-authored 9 publications receiving 528 citations. Previous affiliations of Chi-Yuan Hsu include Sony Broadcast & Professional Research Laboratories.

Rate control for robust video transmission over burst-error wireless channels

Abstract: We study the problem of rate control for transmission of video over burst-error wireless channels, i.e., channels such that errors tend to occur in clusters during fading periods. In particular we consider a scenario consisting of packet based transmission with automatic repeat request (ARQ) error control and a back channel. We start by showing how the delay constraints in real time video transmission can be translated into rate constraints at the encoder, where the applicable rate constraints at a given time depend on future channel rates. With the acknowledgments received through the back channel we have an estimate of the current channel state. This information, combined with an a priori model of the channel, allows us to statistically model the future channel rates. Thus the rate constraints at the encoder can be expressed in terms of the expected channel behavior. We can then formalize a rate distortion optimization problem, namely, that of assigning quantizers to each of the video blocks stored in the encoder buffer such that the quality of the received video is maximized. This requires that the rate constraints be included in the optimization, since violating a rate constraint is equivalent to violating a delay constraint and thus results in losing a video block. We formalize two possible approaches. The first one seeks to minimize the distortion for the expected rate constraints given the channel model and current observation. The second approach seeks to allocate bits so as to minimize the expected distortion for the given model. We use both dynamic programming and Lagrangian optimization approaches to solve these problems. Our simulation results demonstrate that both the video distortion at the decoder and packet loss rate can be significantly reduced when incorporating the channel information provided by the feedback channel and the a priori model into the rate control algorithm.

Joint selection of source and channel rate for VBR video transmission under ATM policing constraints

Abstract: Variable bit-rate (VBR) transmission of video over ATM networks has long been said to provide substantial benefits, both in terms of network utilization and video quality, when compared with conventional constant bit-rate (CBR) approaches. However, realistic VBR transmission environments will certainly impose constraints on the rate that each source can submit to the network. We formalize the problem of optimizing the quality of the transmitted video by jointly selecting the source rate (number of bits used for a given frame) and the channel rate (number of bits transmitted during a given frame interval). This selection is subject to two sets of constraints, namely, (1) the end-to-end delay has to be constant to allow for real-time video display and (2) the transmission rate has to be consistent with the traffic parameters negotiated by user and network. For a general class of constraints, including such popular ones as the leaky bucket, we introduce an algorithm to find the optimal solution to this problem. This algorithm allows us to compare VBR and CBR under the same end-to-end delay constraints. Our results indicate that variable-rate transmission can increase the quality of the decoded sequences without increases in the end-to-end delay. Finally, we show that for the leaky-bucket channel, the channel constraints can be combined with the buffer constraints, such that the system is identical to CBR transmission with an additional, infrequently imposed constraint. Therefore, video quality with a leaky-bucket channel can achieve the same quality of a CBR channel with larger physical buffers, without adding to the physical delay in the system.

Rate control for robust video transmission over wireless channels

Abstract: Video transmission over wireless links is an emerging application which involves a time-varying channel. Compared to other transmission media, wireless links suffer from limited bandwidth, and are more likely to see their performance degrade due to multipath fading. Therefore error control mechanisms, which can achieve better video quality with the available bandwidth and recover from the errors due to the channel degradation, are very important in wireless video transmission systems. Many of the proposed wireless communications systems are likely to be two-way so that a return channel can convey information to the transmitter about the channel state. Recent research has considered ways of improving the transmission reliability by making use of the feedback channel for 'closed loop' error control, including various forms of retransmission. In this paper we propose a rate control algorithm based on dynamic programming combined with automatic repeat request (ARQ) as the error control mechanism. We formalize the constraints imposed by the real-time characteristics of video. We show how when an appropriate model for the channel is available, the overall robustness of the systems can be improved through rate control at the source using the channel state information conveyed by the ARQ acknowledgement.© (1997) COPYRIGHT SPIE--The International Society for Optical Engineering. Downloading of the abstract is permitted for personal use only.

Motion-adaptive de-interlacing method and system for digital televisions

Abstract: One embodiment of the present invention provides a method and system for transforming a video bitstream in an interlaced format into a progressive format which can be displayed by a digital television. For example, the present embodiment utilizes the pixel information of a current field, previous field, and future field of the interlaced video bitstream to try to determine what the original content is of the missing lines of the current field. Specifically, the present embodiment utilizes different sets of pixel information in order to estimate the amount of motion that exist within a video bitstream. In this manner, the present embodiment is able to more closely determine the original value of the missing pixels of each field of the interlaced video bitstream. Therefore, the present embodiment provides a de-interlacing function enabling digital televisions to receive interlaced video bitstreams and display them in the progressive format.

Rate control for robust video transmission over wireless channels

Abstract: Video transmission over wireless links is an emerging application which involves a time-varying channel. Compared to other transmission media, wireless links suffer from limited bandwidth, and are more likely to see theirperformance degrade due to multipath fading. Therefore error control mechanisms, which can achieve better video quality with the available bandwidth and recover from the errors due to the channel degradation, are very important in wireless video transmission systems. Many of the proposed wireless communications systems are likely to be two-way so that a return channel can convey information to the transmitter about the channel state. Recent research has considered ways of improving the transmission reliability by making use of the feedback channel for "closed-loop" error control, including various forms of retransmission. In this paper we propose a rate control algorithm based on dynamic programming combined with Automatic Repeat reQuest (ARQ) as the error control mechanism. We formalize the constraints imposed by the real-time characteristics of video. We show how when an appropriate model for the channel is available, the overall robustness of the system can be improved through rate control at the source using the channel state information conveyed by the ARQ acknowledgement.

Rate-distortion methods for image and video compression

Abstract: In this article 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 framework. We then introduce two popular techniques for resource allocation, namely, Lagrangian optimization and dynamic programming. After a discussion of these techniques as well as some of their extensions, we conclude with a quick review of literature in these areas citing a number of applications related to image and video compression and transmission.

Rate-distortion optimized streaming of packetized media

Abstract: This paper addresses the problem of streaming packetized media over a lossy packet network in a rate-distortion optimized way. We show that although the data units in a media presentation generally depend on each other according to a directed acyclic graph, the problem of rate-distortion optimized streaming of an entire presentation can be reduced to the problem of error-cost optimized transmission of an isolated data unit. We show how to solve the latter problem in a variety of scenarios, including the important common scenario of sender-driven streaming with feedback over a best-effort network, which we couch in the framework of Markov decision processes. We derive a fast practical algorithm for nearly optimal streaming in this scenario, and we derive a general purpose iterative descent algorithm for locally optimal streaming in arbitrary scenarios. Experimental results show that systems based on our algorithms have steady-state gains of 2-6 dB or more over systems that are not rate-distortion optimized. Furthermore, our systems essentially achieve the best possible performance: the operational distortion-rate function of the source at the capacity of the packet erasure channel.

Video coding with optimal inter/intra-mode switching for packet loss resilience

Abstract: Resilience to packet loss is a critical requirement in predictive video coding for transmission over packet-switched networks, since the prediction loop propagates errors and causes substantial degradation in video quality. This work proposes an algorithm to optimally estimate the overall distortion of decoder frame reconstruction due to quantization, error propagation, and error concealment. The method recursively computes the total decoder distortion at pixel level precision to accurately account for spatial and temporal error propagation. The accuracy of the estimate is demonstrated via simulation results. The estimate is integrated into a rate-distortion (RD)-based framework for optimal switching between intra-coding and inter-coding modes per macroblock. The cost in computational complexity is modest. The framework is further extended to optimally exploit feedback/acknowledgment information from the receiver/network. Simulation results both with and without a feedback channel demonstrate that precise distortion estimation enables the coder to achieve substantial and consistent gains in PSNR over known state-of-the-art RD- and non-RD-based mode switching methods.

Transporting real-time video over the Internet: challenges and approaches

Abstract: Delivering real-time video over the Internet is an important component of many Internet multimedia applications. Transmission of real-time video has bandwidth, delay, and loss requirements. However the current Internet does not offer any quality of service (QoS) guarantees to video transmission over the Internet. In addition, the heterogeneity of the networks and end systems makes it difficult to multicast Internet video in an efficient and flexible way. Thus, designing protocols and mechanisms for Internet video transmission poses many challenges. In this paper, we take a holistic approach to these challenges and present solutions from both transport and compression perspectives. With the holistic approach, we design a framework for transporting real-time Internet video, which includes two components, namely, congestion control and error control. Specifically congestion control consists of rate control, rate-adaptive encoding, and rate shaping; error control consists of forward error correction (FEC), retransmission error resilience, and error concealment. For the design of each component in the framework, we classify approaches and summarize representative research work. We point out there exists a design space which can be explored by video application designers and suggest that the synergy of both transport and compression could provide good solutions.

Bit-rate control using piecewise approximated rate-distortion characteristics

Abstract: Digital video's increased popularity has been driven to a large extent by a flurry of international standards (MPEG-1, MPEG-2, H.263, etc). In most standards, the rate control scheme, which plays an important role in improving and stabilizing the decoding and playback quality, is not defined, and thus different strategies can be implemented in each encoder design. Several rate-distortion (R-D)-based techniques have been proposed aimed at the best possible quality for a given channel rate and buffer size. These approaches are complex because they require the R-D characteristics of the input data to be measured before making quantization assignment decisions. We show how the complexity of computing the R-D data can be reduced without significantly reducing the performance of the optimization procedure. We propose two methods which provide successive reductions in complexity by: (1) using models to interpolate the rate and distortion characteristics, and (2) using past frames instead of current ones to determine the models. Our first method is applicable to situations (e.g., broadcast video) where a long encoding delay is possible, while our second approach is more useful for computation-constrained interactive video applications. The first method can also be used to benchmark other approaches. Both methods can achieve over 1 dB peak signal-to-noise rate (PSNR) gain over simple methods like the MPEG Test Model 5 (TM5) rate control, with even greater gains during scene change transitions. In addition, both methods make few a priori assumptions and provide robustness in their performance over a range of video sources and encoding rates. In terms of complexity, our first algorithm roughly doubles the encoding time as compared to simpler techniques (such as TM5). However, the complexity is greatly reduced as compared to methods which exactly measure the R-D data. Our second algorithm has a complexity marginally higher than TM5 and a PSNR performance slightly lower than that of the first approach.