The problem of error control and concealment in video communication is becoming increasingly important because of the growing interest in video delivery over unreliable channels such as wireless networks and the Internet. This paper reviews the techniques that have been developed for error control and concealment in the past ten to fifteen years. These techniques are described in three categories according to the roles that the encoder and decoder play in the underlying approaches. Forward error concealment includes methods that add redundancy at the source end to enhance error resilience of the coded bit streams. Error concealment by postprocessing refers to operations at the decoder to recover the damaged areas based on characteristics of image and video signals. Finally, interactive error concealment covers techniques that are dependent on a dialog between the source and destination. Both current research activities and practice in international standards are covered.

Streaming Video Profile is the subject of an Amendment of MPEG-4, and is developed in response to the growing need on a video-coding standard for streaming video over the Internet. It provides the capability to distribute single-layered frame-based video over a wide range of bit rates with high coding efficiency. It also provides fine granularity scalability (FGS), and its combination with temporal scalability, to address a variety of challenging problems in delivering video over the Internet. This paper provides an overview of the FGS video coding technique in this Amendment of the MPEG-4.

This paper describes the design and analysis of a low-power medium access control (MAC) protocol for wireless/mobile ATM networks. The protocol -- denoted EC-MAC (energy conserving medium access control) -- is designed to support different traffic types with quality-of-service (QoS) provisions. The network is based on the infrastructure model where a base station (BS) serves all the mobiles currently in its cell. A reservation-based approach is proposed, with appropriate scheduling of the requests from the mobiles. This strategy is utilized to accomplish the dual goals of reduced energy consumption and quality of service provision over wireless links. A priority round robin with dynamic reservation update and error compensation scheduling algorithm is used to schedule the transmission requests of the mobiles. Discrete-event simulation has been used to study the performance of the protocol. A comparison of energy consumption of the EC-MAC to a number of other protocols is provided. This comparison indicates the EC-MAC has, in general, better energy consumption characteristics. Performance analysis of the proposed protocol with respect to different quality-of-service parameters using video, audio and data traffic models is provided

fine-grained scalable video over a TCP-friendly connection. The goal is to develop low-complexity yet high-performing schemes that adequately adapt to the short- and long-term variations in available bandwidth. We first present a novel framework for low-complexity streaming of fine-grained scalable video over a TCP-friendly connection. In the context of this scheme, and under the assumption of complete knowledge of bandwidth evolution, we derive an optimal policy for a criterion that involves both image quality and quality variability during playback. Based on this ideal optimal policy, we develop a real-time heuristic to stream fine-grained scalable video over the Internet, and we study its performance using real Internet traces. We find that our heuristic policy performs almost as well as the ideal optimal policy for a wide-range of bandwidth scenarios and when run over ordinary TCP the policy is essentially as good as when running the policy over popular TCP-friendly algorithms.

Real-time audio and video data streamed over unreliable IP networks, such as the Internet, may encounter losses due to dropped packets or late arrivals. This paper reviews error-concealment schemes developed for streaming realtime audio and video data over the Internet. Based on their interactions with (video or audio) source coders, we classify existing techniques into source coder-independent schemes that treat underlying source coders as black boxes, and source coder-dependent schemes that exploit coder-specific characteristics to perform reconstruction. Last, we identify possible future research directions. 1. Introduction Increases in bandwidth and computational speed lead to growing interests in real-time audio and video transmissions over the Internet. In the Internet, packets carrying real-time data may be dropped or arrive too late to be useful because the Internet is a packet-switched, best-effort delivery service, with no guarantee on the quality of service (QoS). Traditi...

Transmission of video over wireless and mobile networks requires a scalable solution that is capable of adapting to the varying channel conditions in real-time (bit-rate scalability). Furthermore, video content needs to be coded in a scalable fashion to match the capabilities of a variety of devices (complexity scalability). These two properties---bit rate and complexity scalability---provide the flexibility that is necessary to satisfy the "Anywhere, Anytime, and Anyone" network paradigm of wireless systems. Meanwhile, MPEG-4 fine-granular-scalability (FGS) has been introduced as a flexible low-complexity solution for video streaming over heterogeneous networks (e.g., the Internet and wireless networks). FGS is also highly resilient to packet losses. However, the flexibility and packet-loss resilience associated with the FGS framework come at the expense of decreased coding efficiency compared with nonscalable coding. In this paper, a novel scalable video-coding framework and corresponding compression methods for wireless video streaming is introduced. Building on the FGS approach, the proposed framework, which we refer to as adaptive motion-compensation FGS (AMC-FGS), provides improved video quality of up to 2 dB. Furthermore, the new scalability structures provide the FGS framework with the flexibility to provide tradeoffs between resilience, higher coding efficiency and terminal complexity for more efficient wireless transmission.

Video transmission over the current best-effort Internet should be made fair to com- peting TCP traffic. Because the available bandwidth to a TCP-friendly stream changes significantly over medium and long time scales, it is desirable to adapt the streaming rate of the video to the current bandwidth conditions. We consider two adaptive streaming schemes for stored video. The first scheme switches among multiple encoded versions, with each version encoded at a different rate. The second scheme adds and drops encoding layers. To compare the two schemes, we develop streaming control policies for each scheme and evaluate their performance using trace-driven simulation. Our results show that when analogous streaming policies are used, switching versions outperforms adding/dropping layers because of the overhead associated with layering. However, the enhanced flexibility of layering can compensate the performance degradation due to the layering overhead.

This paper describes the design and analysis of the scheduling algorithm for EC-MAC (energy conserving medium access control) [1], a low-power medium access control (MAC) protocol for wireless and mobile ATM networks. Based on the structure of EC-MAC and the characteristics of wireless channel, we propose a new algorithm which can deal with the bursty errors and the location-dependent errors. Most scheduling algorithms proposed for either wired or wireless networks were analyzed with homogeneous traffic or multimedia services with simplified traffic models. We analyze our scheduling algorithm with more realistic multimedia traffic models. One of the key goals of the scheduling algorithm is simplicity and fast implementation. Unlike the time-stamp based algorithm, our algorithm does not need to sort the virtual time, thus reducing the complexity of the algorithm significantly.

Several embedded video coding schemes have been recently developed for multimedia streaming over IP. In particular, fine-granular-scalability (FGS) video coding has been recently adopted by the MPEG-4 standard as the core video-compression method for streaming applications. From its inception, the FGS scalability structure was designed to be packet-loss resilient especially under unequal packet-loss protection (UPP). However, since the introduction of FGS, there has not been a comprehensive study evaluating its packet-loss resilience under unrecoverable packet losses that are common in Internet streaming applications. In this paper, we evaluate two important aspects of FGS packet-loss resilience. First, we study the impact of applying UPP between the base- and enhancement-layers on FGS-based streams, and we compare equal packet-loss protection (EPP) with UPP scenarios. Second, we introduce the notion of fine-grained loss protection (FGLP), which is suitable for the FGS enhancement-layer, and we develop an analytical framework for evaluating FGLP bounds. Based on these bounds, we show the impact of applying fine-grained protection to the FGS enhancement-layer for different types of video sequences and over a wide range of bit-rates and packet-loss ratios. As illustrated by our extensive simulation results, applying 1) UPP between the base- and enhancement-layers and 2) FGLP for the FGS enhancement-layer can provide significant resilience under moderate-to-high packet-loss ratios (e.g., 5--10%). Furthermore, the merits of this new packet-loss protection technique go beyond the FGS coding scheme, because FGLP can be successfully applied to improve the resilience to packet-losses of other embedded video coding techniques.

In this paper, we develop an approach toward joint source-channel coding for motion-compensated DCT-based scalable video coding and transmission. A framework for the optimal selection of the source and channel coding rates over all scalable layers is presented such that the overall distortion is minimized. The algorithm utilizes universal rate distortion characteristics which are obtained experimentally and show the sensitivity of the source encoder and decoder to channel errors. The proposed algorithm allocates the available bit rate between scalable layers and, within each layer, between source and channel coding. We present the results of this rate allocation algorithm for video transmission over a wireless channel using the H.263 Version 2 signal-to-noise ratio (SNR) scalable codec for source coding and rate-compatible punctured convolutional (RCPC) codes for channel coding. We discuss the performance of the algorithm with respect to the channel conditions, coding methodologies, layer rates, and number of layers.