Joint QoS control for video streaming over wireless multihop networks: A cross-layer approach

In this paper, we propose a novel cross-layer framework for jointly controlling and coding for multiple video streams in wireless multihop networks. At first, we develop a cross-layer flow control algorithm that works at the medium access control (MAC) layer to adjust each link's persistence probability and at the transport layer to adjust flow rates. This proposal is designed in a distributed manner that is amenable to online implementation for wireless networks, and then, a rate-distortion optimized joint source-channel coding (JSCC) approach for error-resilient scalable encoded video is presented, in which the video is encoded into multiple independent streams and each stream is assigned forward error correction (FEC) codes to avoid error propagation. Furthermore, we integrate the JSCC with the specific flow control algorithm, which optimally applies the appropriate channel coding rate given the constraints imposed by the transmission rate obtained from the proposed flow control algorithm and the prevailing channel condition. Simulation results demonstrate the merits and the need for joint quality of service (QoS) control in order to provide an efficient solution for video streaming over wireless multihop networks.     

Packet-by-Packet Adaptive Scheme for Video Coding and Transmission Over Wireless IP Networks Using Rate-Compatible Punctured Turbo (RCPT) Codes

In this paper, we consider real-time video coding and transmission over packet-switched wireless IP networks, such as WLAN, using RCPT codes and joint source-channel coding (JSCC) with concentration on a packet-by-packet adaptive scheme. We present a systematic design methodology to enable the applicability of JSCC techniques. The performance of H.263+ video coding and transmission over wireless channel modeled as slow Rician fading channels using this approach is studied. Results indicate that a packet-by-packet adaptive RCPT-JSCC approach is of significant advantage for real-time video applications and leads to more acceptable video delivery quality over interference-limited and time-varying wireless networks.     

Robust Packet Video Transport Over Wireless Fading Channels Using a Joint Source-channel Coding Approach

In this paper, we address the problem of robust video transmission over wireless networks. Specifically, we consider packet video transmission over wireless IP networks based on the RTP/UDP/IP protocol stack. Digital video delivered over wireless networks is expected to suffer quality degradation from both packet loss and bit errors in the payload. In this paper, both packet loss and bit errors in the payload are considered and the performance of a joint source-channel coding (JSCC) approach employing forward error-correction (FEC) coding schemes for H.263 +  video transmission is studied. Results indicate that with an appropriate JSCC approach, FEC-based error-control techniques can significantly improve the packetization efficiency for a given end-to-end quality requirement and lead to more acceptable video delivery quality over time-varying wireless networks. Another important observation is that with a JSCC approach the fading effects of wireless links upon end-to-end video quality are substantially decreased compared to a system without using channel coding, resulting in attractive robust performance characteristics.

Cross-layer architecture for scalable video transmission in wireless network

Multimedia applications such as video conference, digital video broadcasting (DVB), and streaming video and audio have been gaining popularity during last years and the trend has been to allocate these services more and more also on mobile users. The demand of quality of service (QoS) for multimedia raises huge challenges on the network design, not only concerning the physical bandwidth but also the protocol design and services. One of the goals for system design is to provide efficient solutions for adaptive multimedia transmission over different access networks in all-IP environment. The joint source and channel coding (JSCC/D) approach has already given promising results in optimizing multimedia transmission. However, in practice, arranging the required control mechanism and delivering the required side information through network and protocol stack have caused problems and quite often the impact of network has been neglected in studies. In this paper we propose efficient cross-layer communication methods and protocol architecture in order to transmit the control information and to optimize the multimedia transmission over wireless and wired IP networks. We also apply this architecture to the more specific case of streaming of scalable video streams. Scalable video coding has been an active research topic recently and it offers simple and flexible solutions for video transmission over heterogeneous networks to heterogeneous terminals. In addition it provides easy adaptation to varying transmission conditions. In this paper we illustrate how scalable video transmission can be improved with efficient use of the proposed cross-layer design, adaptation mechanisms and control information.     

A joint source-channel distortion model for JPEG compressed images.

The need for efficient joint source-channel coding (JSCC) is growing as new multimedia services are introduced in commercial wireless communication systems. An important component of practical JSCC schemes is a distortion model that can predict the quality of compressed digital multimedia such as images and videos. The usual approach in the JSCC literature for quantifying the distortion due to quantization and channel errors is to estimate it for each image using the statistics of the image for a given signal-to-noise ratio (SNR). This is not an efficient approach in the design of real-time systems because of the computational complexity. A more useful and practical approach would be to design JSCC techniques that minimize average distortion for a large set of images based on some distortion model rather than carrying out per-image optimizations. However, models for estimating average distortion due to quantization and channel bit errors in a combined fashion for a large set of images are not available for practical image or video coding standards employing entropy coding and differential coding. This paper presents a statistical model for estimating the distortion introduced in progressive JPEG compressed images due to quantization and channel bit errors in a joint manner. Statistical modeling of important compression techniques such as Huffman coding, differential pulse-coding modulation, and run-length coding are included in the model. Examples show that the distortion in terms of peak signal-to-noise ratio (PSNR) can be predicted within a 2-dB maximum error over a variety of compression ratios and bit-error rates. To illustrate the utility of the proposed model, we present an unequal power allocation scheme as a simple application of our model. Results show that it gives a PSNR gain of around 6.5 dB at low SNRs, as compared to equal power allocation.     

Cross-layer optimized authentication and error control for wireless 3D medical video streaming over LTE

3D video for tele-medicine applications is gradually gaining momentum since the 3D technology can provide precise location information. However, the weak link for 3D video streaming is the necessary wireless link of the communication system. Neglecting the wireless impairments can severely degrade the performance of 3D video streaming that communicates complex critical medical data. In this paper, we propose systematic methodology for ensuring high performance of the 3D medical video streaming system. First, we present a recursive end-to-end distortion estimation approach for MVC (multiview video coding)-based 3D video streaming over error-prone networks by considering the 3D inter-view prediction. Then, based on the previous model, we develop a cross-layer optimization scheme that considers the LTE wireless physical layer (PHY). In this optimization, the authentication requirements of 3D medical video are also taken into account. The proposed cross-layer optimization approach jointly controls and manages the authentication, video coding quantization of 3D video, and the modulation and channel coding scheme (MCS) of the LTE wireless PHY to minimize the end-to-end video distortion. Experimental results show that the proposed approach can provide superior 3D medical video streaming performance in terms of peak signal-to-noise ratio (PSNR) when compared to state-of-the-art approaches that include joint source-channel optimized streaming with multi-path hash-chaining based-authentication, and also conventional video streaming with single path hash-chaining-based authentication.     

Rate-distortion optimized hybrid error control for real-time packetized video transmission.

The problem of application-layer error control for real-time video transmission over packet lossy networks is commonly addressed via joint source-channel coding (JSCC), where source coding and forward error correction (FEC) are jointly designed to compensate for packet losses. In this paper, we consider hybrid application-layer error correction consisting of FEC and retransmissions. The study is carried out in an integrated joint source-channel coding (IJSCC) framework, where error resilient source coding, channel coding, and error concealment are jointly considered in order to achieve the best video delivery quality. We first show the advantage of the proposed IJSCC framework as compared to a sequential JSCC approach, where error resilient source coding and channel coding are not fully integrated. In the USCC framework, we also study the performance of different error control scenarios, such as pure FEC, pure retransmission, and their combination. Pure FEC and application layer retransmissions are shown to each achieve optimal results depending on the packet loss rates and the round-trip time. A hybrid of FEC and retransmissions is shown to outperform each component individually due to its greater flexibility.     

Multipath video transport over ad hoc networks

Real-time multimedia transport has stringent bandwidth, delay, and loss requirements. It is a great challenge to support such applications in wireless ad hoc networks, which are characterized by frequent link failures and congestion. Using multiple paths in parallel for a real-time multimedia session (called multipath transport) provides a new degree of freedom in designing robust multimedia transport systems. In this article, we describe a framework for multipath video transport over wireless ad hoc networks, and examine its essential components, including multistream video coding, multipath routing, and transport mechanisms. We illustrate by three representative examples how to extend existing video coding schemes in order to fully explore the potential of multipath transport. We also examine important mechanisms in different layers for supporting multipath video transport over ad hoc networks. Our experiments show that multipath transport is a promising technique for efficient video communications over ad hoc networks.     

Forward error control for MPEG-2 video transport in a wireless ATM LAN

The possibility of providing multimedia services to mobile users has led to interest in designing broadband wireless networks that can guarantee quality of service for traffic flows. However, a fundamental problem in these networks is that severe losses may occur due to the random fading characteristics of the wireless channel. Error control algorithms which compensate for these losses are required in order to achieve reasonable loss rates. In this paper, the performance of error control based on forward error correction (FEC) for MPEG-2 video transmission in an indoor wireless ATM LAN is studied. A random bit error model and a multipath fading model are used to investigate the effect of errors on video transport. Combined source and channel coding techniques that employ single-layer and scalable MPEG-2 coding to combat channel errors are compared. Simulation results indicate that FEC-based error control in combination with 2-layer video coding techniques can lead to acceptable quality for indoor wireless ATM video.Work performed while the author was at AT&T Bell Laboratories on a D.O.E. fellowship program.     

Joint Source-Channel Distortion Modeling for MPEG-4 Video

Multimedia communication has become one of the main applications in commercial wireless systems. Multimedia sources, mainly consisting of digital images and videos, have high bandwidth requirements. Since bandwidth is a valuable resource, it is important that its use should be optimized for image and video communication. Therefore, interest in developing new joint source-channel coding (JSCC) methods for image and video communication is increasing. Design of any JSCC scheme requires an estimate of the distortion at different source coding rates and under different channel conditions. The common approach to obtain this estimate is via simulations or operational rate-distortion curves. These approaches, however, are computationally intensive and, hence, not feasible for real-time coding and transmission applications. A more feasible approach to estimate distortion is to develop models that predict distortion at different source coding rates and under different channel conditions. Based on this idea, we present a distortion model for estimating the distortion due to quantization and channel errors in MPEG-4 compressed video streams at different source coding rates and channel bit error rates. This model takes into account important aspects of video compression such as transform coding, motion compensation, and variable length coding. Results show that our model estimates distortion within 1.5 dB of actual simulation values in terms of peak-signal-to-noise ratio.      

A New Transport Protocol for Broadcasting/Multicasting MPEG-2 Video Over Wireless ATM Access Networks

Because of the telecommunications de-regulation act and progress in wireless technologies, we will see the co-existence of heterogeneous broadband access infrastructures in the broadband video service industry in the near future. In this paper, we addressed the error control issue when transmitting MPEG-2 video streams over wireless access networks for broadband video broadcast or multicast services. An end-to-end transport protocol based on ATM and wireless ATM technologies is proposed. For video services, the underlying transport network should be transparent and quality should be maintained uniformly over all the segments whether wireline or wireless links. For network resources to be used efficiently, error control should be applied locally on the wireless segments so as to avoid the excessive overhead over the reliable wireline portions. Because a broadband video broadcast or multicast service is a one-to-multiple point service, FEC is the most prevalent error control mechanism. Due to the important role of MPEG-2 control information in the decoding process, priority MPEG-2 control information has to be differentiated from MPEG-2 data information, and excess error protection has to be allocated to it in order to achieve satisfactory QoS. Therefore, a header redundancy FEC (HRFEC) scheme for error control is applied at the local distribution centers before the MPEG-2 encoded video streams are transmitted over the wireless channels. HRFEC is an FEC-based selective protection scheme, which allocates extra error protection to important control information. Simulation results show that the quality of the reconstructed video sequence is vastly improved by using HRFEC, when the channel condition is poor.     

Feedback-based error control for mobile video transmission

We review feedback-based low bit-rate video coding techniques for robust transmission in mobile multimedia networks. For error control on the source coding level, each decoder has to make provisions for error detection, resynchronization, and error concealment, and we review techniques suitable for that purpose. Further, techniques are discussed for intelligent processing of acknowledgment information by the coding control to adapt the source coder to the channel. We review and compare error tracking, error confinement, and reference picture selection techniques for channel-adaptive source coding. For comparison of these techniques, a system for transmitting low bit-rate video over a wireless channel is presented and the performance is evaluated for a range of transmission conditions. We also show how feedback-based source coding can be employed in conjunction with precompressed video stored on a media server. The techniques discussed are applicable to a wide variety of interframe video schemes, including various video coding standards. Several of the techniques have been incorporated into the H.263 video compression standard, and this standard is used as an example throughout     

End-to-end optimized TCP-friendly rate control for real-time video streaming over wireless multi-hop networks

Rate control is an important issue in video streaming applications. The most popular rate control scheme over wired networks is TCP-Friendly Rate Control (TFRC), which is designed to provide optimal transport service for unicast multimedia delivery based on the TCP Reno’s throughput equation. It assumes perfect link quality, treating network congestion as the only reason for packet losses. Therefore, when used in wireless environment, it suffers significant performance degradation because of packet losses arising from time-varying link quality. Most current research focuses on enhancing the TFRC protocol itself, ignoring the tightly coupled relation between the transport layer and other network layers. In this paper, we propose a new approach to address this problem, integrating TFRC with the application layer and the physical layer to form a holistic design for real-time video streaming over wireless multi-hop networks. The proposed approach can achieve the best user-perceived video quality by jointly optimizing system parameters residing in different network layers, including real-time video coding parameters at the application layer, packet sending rate at the transport layer, and modulation and coding scheme at the physical layer. The problem is formulated and solved as to find the optimal combination of parameters to minimize the end-to-end expected video distortion constrained by a given video playback delay, or to minimize the video playback delay constrained by a given end-to-end video distortion. Experimental results have validated 2–4 dB PSNR performance gain of the proposed approach in wireless multi-hop networks by using H.264/AVC and NS-2.     

End-to-end TCP-friendly streaming protocol and bit allocation for scalable video over wireless Internet

With the convergence of wired-line Internet and mobile wireless networks, as well as the tremendous demand on video applications in mobile wireless Internet, it is essential to an design effective video streaming protocol and resource allocation scheme for video delivery over wireless Internet. Taking both network conditions in the Internet and wireless networks into account, in this paper, we first propose an end-to-end transmission control protocol (TCP)-friendly multimedia streaming protocol for wireless Internet, namely WMSTFP, where only the last hop is wireless. WMSTFP can effectively differentiate erroneous packet losses from congestive losses and filter out the abnormal round-trip time values caused by the highly varying wireless environment. As a result, WMSTFP can achieve higher throughput in wireless Internet and can perform rate adjustment in a smooth and TCP-friendly manner. Based upon WMSTFP, we then propose a novel loss pattern differentiated bit allocation scheme, while applying unequal loss protection for scalable video streaming over wireless Internet. Specifically, a rate-distortion-based bit allocation scheme which considers both the wired and the wireless network status is proposed to minimize the expected end-to-end distortion. The global optimal solution for the bit allocation scheme is obtained by a local search algorithm taking the characteristics of the progressive fine granularity scalable video into account. Analytical and simulation results demonstrate the effectiveness of our proposed schemes.     

无线多媒体传感器网络视频编码研究

在资源受限的无线多媒体传感器网络中存在大量的视频数据需要处理和传输.为充分地利用这些有限的资源,需要设计一个能够综合考虑能量消耗和压缩率、图像质量之间平衡的视频编码方案.对无线多媒体传感器网络中视频编码技术的发展现状、面临的挑战和设计目标进行了讨论.对无线多媒体传感器网络视频编码现有的解决方案和理论研究成果进行探讨,分...     

Ad Hoc上的新协议——流实时传输协议

在Ad Hoc网络中传输多媒体数据会受到网络拓扑变化和无线信道干扰的限制.多流实时传输协议MRTP(multi-flow real-time transport protocol)是一种可以分配和传输多个视频数据流的新协议.它结合多描述编码和多径传输,能提高无线对等网络上视频传输的效率和质量.     

Joint source-channel coding for wireless object-based video communications utilizing data hiding.

In recent years, joint source-channel coding for multimedia communications has gained increased popularity. However, very limited work has been conducted to address the problem of joint source-channel coding for object-based video. In this paper, we propose a data hiding scheme that improves the error resilience of object-based video by adaptively embedding the shape and motion information into the texture data. Within a rate-distortion theoretical framework, the source coding, channel coding, data embedding, and decoder error concealment are jointly optimized based on knowledge of the transmission channel conditions. Our goal is to achieve the best video quality as expressed by the minimum total expected distortion. The optimization problem is solved using Lagrangian relaxation and dynamic programming. The performance of the proposed scheme is tested using simulations of a Rayleigh-fading wireless channel, and the algorithm is implemented based on the MPEG-4 verification model. Experimental results indicate that the proposed hybrid source-channel coding scheme significantly outperforms methods without data hiding or unequal error protection.