Transmission distortion estimation for real-time video delivery over hybrid channels with bit errors and packet erasures

We study transmission distortion of video delivery over hybrid channels with bit errors and packet erasures. Hybrid channels have been prone to suffering from transmission error in the forms of packet erasures due to congestion at wired networks, as well as bit errors caused by the wireless interference or fading on wireless links. We present a recursion approach to transmission distortion estimation over hybrid channels in successive frames with the prediction mode. This approach is feasible for video transmission applications as it is capable of online estimation of transmission distortion on received decoded video. Our extensive simulation results on six well-known video sequences demonstrate that our proposed approach is accurate and robust.     

Error robust scalable audio streaming over wireless IP networks

Streaming high-fidelity audio over wireless Internet protocol (IP) networks is a challenging task because the networks present not only packet losses, but also residual bit errors. These losses and errors have severe adverse effect on the compressed audio bitstream. To solve this problem, this paper introduces error resilience in conjunction with error protection for scalable audio streaming over wireless networks. Specifically, error resilience is achieved by performing bitstream data partitioning and reversible variable length coding in the audio coder. Error protection is provided by layered product channel code to simultaneously handle packet losses and residual bit errors. Both the row and column codes of the product code provide unequal error protection for different layers of the audio bitstream by considering the characteristics of the scalable audio. Rate-distortion optimization is performed to determine the best source-channel coding tradeoff that minimizes the average expected end-to-end distortion. Simulation results demonstrate the effectiveness of our proposed approach.     

一种标准兼容的多描述图像编码分析及改进

多描述编码被证明是用于非优先级网络中传输多媒体数据的有效方法,而现有很多算法与图像标准不兼容,分析了一种与标准兼容的多描述图像编码算法,该算法可以提供灵活冗余插入,以提高丢包或误码环境下的图像传输性能。通过深入分析该算法在不同比特率下的表现,发现其在低比特率端性能与较高比特率下的表现迥异,结合区分对待的思想来改善了这一缺点,并利用JPEG2000的ROI特性证明其有效性。     

Cross-Layer prioritized H.264 video packetization and error protection over noisy channels

Video transmission over wireless channels is affected by channel-induced packet losses. Distortion due to channel errors can be alleviated by applying forward error correction. Aggregating H.264/AVC slices to form video packets with sizes adapted to their importance can also improve transmission reliability. Larger packets are more likely to be in error but smaller packets require more overhead. We present a cross-layer dynamic programming (DP) approach to minimize the expected received video distortion by jointly addressing the priority-adaptive packet formation at the application layer and rate compatible punctured convolutional (RCPC) code rate allocation at the physical layer for prioritized slices of each group of pictures (GOP). Some low priority slices are also discarded to improve protection to more important slices and meet the channel bit-rate limitations. We propose two schemes. Our first scheme carries out joint optimization for all slices of a GOP at a time. The second scheme extends our cross-layer DP-based approach to slices of each frame by predicting the expected channel bit budget per frame for live streaming. The prediction uses a generalized linear model developed over the cumulative mean squared error per frame, channel SNR, and normalized compressed frame bit budget. The parameters are determined over a video dataset that spans high, medium and low motion complexity. The predicted frame bit budget is used to derive the packet sizes and corresponding RCPC code rates for live transmission using our DP-based approach. Simulation results show that both proposed schemes significantly improve the received video quality over contemporary error protection schemes.     

Hybrid Erasure-Error Protocols for Wireless Video

Many recently proposed cross-layer protocols for wireless video, have advocated the relay of corrupted packet to higher layers. Such protocols lead to both errors and erasures at the compressed video application layer. We generically refer to such schemes as hybrid erasure-error protocols (HEEPs). In this paper, we analyze the utility of HEEPs for efficient transmission of video over wireless channels. In order to maintain the generic nature of the deductions in this paper, we base our analysis on two (rather abstract) communication schemes for wireless video: hybrid error-erasure cross-layer design (CLD) and hybrid error-erasure cross-layer design with side-information (CLDS). We make a comparative analysis of the channel capacities of these schemes over single and multi-hop wireless channels to identify the conditions under which the HEEPs can provide improved performance over conventional (CON) protocols. In addition, we employ Reed Solomon (RS) and low-density parity check (LDPC)-code-based forward-error correction (FEC) schemes to illustrate that the improvement in capacity can easily enable an FEC scheme employed in conjunction with a HEEP to provide improved throughput. Finally we compare the performance of CON, CLD, and CLDS in terms of video quality using the H.264 video standard. The simulation results show a significant advantage for the HEEPs     

Efficient channel code rate selection algorithms for forward error correction of packetized multimedia bitstreams in varying channels

We study joint source-channel coding systems for the transmission of images over varying channels without feedback. We consider the situation where the channel statistics are unknown to the transmitter and focus on systems that enable good performance over a wide range of channel conditions. We first propose a linear-time channel code rate selection algorithm for a hybrid transmission system that combines packetization of an embedded wavelet bitstream into independently decodable packets and forward error correction with a concatenated cyclic redundancy check/rate-compatible punctured convolutional (RCPC) channel coder. We then consider an extension of this hybrid system with additional Reed-Solomon (RS) coding across the packets and give a linear-time algorithm for the efficient selection of both the RS and RCPC code rates. Experimental results for a wireline/wireless link modeled as the combination of a packet erasure channel and a Rayleigh flat-fading channel showed that our schemes significantly outperformed the best previous forward error correction systems in many situations where the actual channel parameter values deviated from the ones used in the optimization of the source-channel rate allocation.     

多描述编码技术的新发展

随着通信与视频编码技术的发展，通过网络与无线信道传输视频内容成为可能，但在这些应用环境下，视频内容会因为丢包(网络)或信道失效(无线信道)而造成数据丢失。解决信道传输错误的问题可以从信源编码、信道编码及信源信道联合编码3种途径进行改进。如今多描述编码已被证明是克服传输错误，尤其是丢包的一种有效的信源编码方法。为了使人们对多描述编码研究的现状有一概括了解，首先介绍了多描述编码的主要特征；然后分别从信源编码及信道编码的角度归纳了及现有解决丢包问题的主要方法；接下来深入讨论了多描述编码的应用与研究的新进展；最后指出了多描述编码发展的方向。     

Decoding of random network codes

We consider the decoding for Silva-Kschischang-Kötter random network codes based on Gabidulin’s rank-metric codes. The model of a random network coding channel can be reduced to transmitting matrices of a rank code through a channel introducing three types of additive errors. The first type is called random rank errors. To describe other types, the notions of generalized row erasures and generalized column erasures are introduced. An algorithm for simultaneous correction of rank errors and generalized erasures is presented. An example is given.     

Optimal Packet Loss Protection of Progressively Compressed 3-D Meshes

We consider a state-of-the-art system that uses layered source coding and forward error correction with Reed-Solomon codes to efficiently transmit 3-D meshes over lossy packet networks. Given a transmission bit budget, the performance of this system can be optimized by determining how many layers should be sent, how each layer should be packetized, and how many parity bits should be allocated to each layer such that the expected distortion at the receiver is minimum. The previous solution for this optimization problem uses exhaustive search, which is not feasible when the transmission bit budget is large. We propose instead an exact algorithm that solves this optimization problem in linear time and space. We illustrate the advantages of our approach by providing experimental results for the compressed progressive meshes (CPM) mesh compression technique.     

可伸缩多媒体传输错误保护算法综述

可伸缩多媒体对传输中的丢包和误码极为敏感,需要有效的错误保护．主要讨论基于前向纠错的错误保护算法,其核心是利用率失真优化和信源信道联合编码等技术在信源和信道之间合理分配码率,以使接收方重构信号质量最好．依据信道的错误统计特性,将传输信道分为丢包信道和无线误码信道两种,对每种信道模型的错误保护模式和优化目标都给出了统一的形式化问题描述,对主流算法的技术特点和性能指标等进行了分析和比较,最后讨论了下一步研究方向．     

精细分级视频码流传输和容错性能的研究

精细分级编码是解决网络异构性的有效方法之一。编码输出码流的容错性能是衡量编码技术性能的重要指标。该文实现了一个支持精细分级编码的H.264编解码器,并对精细分级码流在IP网上的传输性能进行了研究。提出了一种精细分级码流容错传输打包策略。实验结果表明,该方案具有良好的容错性能。     

基于RS码的网络编码层设计

针对高丢包率的无线网络环境下TCP/IP协议由于其自身缺陷难以提高网络吞吐量的问题,在TCP/IP层次结构中的传输层和网络层之间引入了一个网络编码层。此层在有线网络环境下,对于传输层和网络层来说是透明的,不影响TCP/IP层次结构的功能;在高丢包率的网络环境下,通过编码窗口使用基于RS码的网络编码来提高网络传输吞吐量。网络编码层模型被引入NS2网络仿真软件,仿真结果表明,在高丢包率的网络环境下,此层可提高网络传输吞吐量。     

基于小波EBCOT的图像IP网络传输控制策略

为了使当前“尽力而为”的网络提供视频流服务时满足QoS要求，文章提出一种基于小波EBCOT的图像IP网络传输控制策略．通过采用基于小波EBCOT的渐进可分级编码方法，对压缩后的比特流按其重要性分层打包传输，同时根据对当前网络可用带宽的估计及信道状态的判断，区分网络拥塞及不可靠传输两种不同情况进行自适应不等重丢包保护AUPLP．软件仿真表明，该文算法可大大增强小波EBCOT编码后数据的抗误码能力，在发生数据拥塞时有助于缓解网络的过负载状况，在发生不可靠传输时接收端解码图像能平均提高1．2dB的PSNR。     

Joint source coding and packet classification for real-time video transmission over differentiated services networks

Differentiated Services (DiffServ) is one of the leading architectures for providing quality of service in the Internet. We propose a scheme for real-time video transmission over a DiffServ network that jointly considers video source coding, packet classification, and error concealment within a framework of cost-distortion optimization. The selections of encoding parameters and packet classification are both used to manage end-to-end delay variations and packet losses within the network. We present two dual formulations of the proposed scheme: the minimum distortion problem, in which the objective is to minimize the end-to-end distortion subject to cost and delay constraints, and the minimum cost problem, which minimizes the total cost subject to end-to-end distortion and delay constraints. A solution to these problems using Lagrangian relaxation and dynamic programming is given. Simulation results demonstrate the advantage of jointly adapting the source coding and packet classification in DiffServ networks.     

Providing adaptive QoS to layered video over wireless local area networks through real-time retry limit adaptation

Robust streaming of video over 802.11 wireless LANs (WLANs) poses many challenges, including coping with packets losses caused by network buffer overflow or link erasures. In this paper, we propose a novel error protection method that can provide adaptive quality-of-service (QoS) to layered coded video by utilizing priority queueing at the network layer and retry-limit adaptation at the link layer. The design of our method is motivated by the observation that the retry limit settings of the MAC layer can be optimized in such a way that the overall packet losses that are caused by either link erasure or buffer overflow are minimized. We developed a real-time retry limit adaptation algorithm to trace the optimal retry limit for both the single-queue (or single-layer) and multiqueue (or multilayer) cases. The video layers are unequally protected over the wireless link by the MAC with different retry limits. In our proposed transmission framework, these retry limits are dynamically adapted depending on the wireless channel conditions and traffic characteristics. Furthermore, the proposed priority queueing discipline is enhanced with packet filtering and purging functionalities that can significantly save bandwidth by discarding obsolete or un-decodable packets from the buffer. Simulations show that the proposed cross-layer protection mechanism can significantly improve the received video quality.     

Robust capacity maximization transceiver design for MIMO OFDM systems多天线正交频分复用系统中鲁棒性容量最大化的收发机设计

In this paper, we investigated capacity maximization problem for Multiple Input Multiple Output Orthogonal Frequency Division Multiplexing systems with imperfect channel state information (CSI). To the best of our knowledge, the considered problem is still an open problem. However, the transceiver designs for MIMO OFDM systems have been extensively studied. It seems nobody gives closed-form solutions for resource allocation for MIMO OFDM systems with statistical channel estimation errors up to date. In our work, based on practical channel estimation algorithm, the channel estimation errors are first derived and then the robust resource allocation problem has been formulated. The structure of the optimal robust precoder is first derived, based on which the optimization problem will be simplified significantly. Furthermore, based on the Lagrangian dual method, a robust power allocation algorithm is proposed. The proposed power allocation can be considered as a variant of water-filling solution named cluster water-filling solution. Finally, simulation results show that our proposed robust design outperforms the non-robust design in terms of channel capacity.     

无线网络中一种智能控制的组播拥塞控制机制

针对现有组播拥塞控制算法应用到无线网络中存在的性能下降问题,提出一种基于新的智能组播拥塞控制机制ECMCC。ECMCC机制根据网络相对队列时延和数据包丢失检测网络的拥塞状态,采用代表集合机制反馈信息,利用专家控制器的推理判断区分丢包原因和当前的网络状态,进而采取不同的控制策略调节组播源端发送速率。仿真结果表明,ECMCC机制收敛速度快、灵敏性好、速率变化平滑,在有线网络中具有良好的TCP友好性。同时,ECMCC能有效区分网络拥塞和随机差错,提高了网络的吞吐量,适用于无线网络环境,且在无线网络较低误码率时具有一定的TCP友好性。     

Symbol-level reliable broadcasting of sensitive data in error-prone wireless networks

Reliable packet transmission over error-prone wireless networks has received a lot of attention from the research community. In this paper, instead of using simple packet retransmissions to provide reliability, we consider a novel retransmission approach, which is based on the importance of bits (symbols). We study the problem of maximizing the total gain in the case of partial data delivery in error-prone wireless networks, in which each set of bits (called symbols) has a different weight. We first address the case of one-hop single packet transmission, and prove that the optimal solution that maximizes the total gain has a round-robin symbol transmission pattern. Then, we extend our solution to the case of multiple packets. We also enhance the expected gain using random linear network coding. Our simulation results show that our proposed multiple packets transmission mechanism can increase the gain up to 60%, compared to that of a simple retransmission. Moreover, our network coding scheme enhances the expected total gain up to 15%, compared to our non-coding mechanism.     

Erasure correction by low-density codes

We generalize the method for computing the number of errors correctable by a low-density parity-check (LDPC) code in a binary symmetric channel, which was proposed by V.V. Zyablov and M.S. Pinsker in 1975. This method is for the first time applied for computing the fraction of guaranteed correctable erasures for an LDPC code with a given constituent code used in an erasure channel. Unlike previously known combinatorial methods for computing the fraction of correctable erasures, this method is based on the theory of generating functions, which allows us to obtain more precise results and unify the computation method for various constituent codes of a regular LDPC code. We also show that there exist an LDPC code with a given constituent code which, when decoded with a low-complexity iterative algorithm, is capable of correcting any erasure pattern with a number of erasures that grows linearly with the code length. The number of decoding iterations, required to correct the erasures, is a logarithmic function of the code length. We make comparative analysis of various numerical results obtained by various computation methods for certain parameters of an LDPC code with a constituent single-parity-check or Hamming code.     

Analysis of link-level hybrid FEC/ARQ-SR for wireless links and long-lived TCP traffic

Since the TCP protocol uses the loss of packets as an indication of network congestion, its performance degrades over wireless links, which are characterized by a high bit error rate. Different solutions have been proposed to improve the performance of TCP over wireless links, the most promising one being the use of a hybrid model at the link level combining Forward Error Correction (FEC), Automatic Repeat Request with Selective Repeat (ARQ-SR), and an in-order delivery of packets to IP. The drawback of FEC is that it consumes some extra bandwidth to transmit the redundant information. ARQ-SR consumes extra bandwidth only when packets are lost, its drawback is that it increases the round-trip time (RTT), which may deteriorate the performance of TCP. Another drawback of ARQ-SR is that a complete packet can be retransmitted to correct a small piece of errored data. We study in this paper the performance of TCP over a wireless link implementing hybrid FEC/ARQ-SQ. The study is done by simulating and modeling long-lived TCP transfers over wireless links showing Bernoulli errors. We are motivated by how to tune link-level error recovery, e.g. amount of FEC, persistency of ARQ, so as to maximize the performance of TCP. We provide results for different physical characteristics of the wireless link (delay, error rate) and for different traffic loads (number of TCP connections).