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.     

基于RCPC-FEC的3G视频传输模型研究

为了有效克服3G无线网络传输中的比特错误和数据包丢失问题,采用码率兼容删除卷积码（RCPC）抑制比特错误问题,通过前向纠错（FEC）减小数据丢包率,将RCPC与FEC有机结合提出了一种基于RCPC-FEC的无线网络视频传输模型。在3G无线网络传输的不同比特错误率和数据丢包率下对该模型进行了测试,实验结果证明了模型的有效性和可行性。     

基于TCP友好速率控制和前向纠错的MPEG-2视频传输

针对Internet视频传输面临拥塞控制和数据包丢失的问题，结合TCP友好的速率控制算法和前向纠错机制建立视频传输的分层体系构架和控制策略。传输体系同时采用以GOP为基本分析单元的视频帧速率预测模型，实现根据网络丢包率的变化动态地优化配置前向纠错的冗余信息。实验证明，传输体系采用动态优化的前向纠错能实时地适应带宽的变化，有效地降低数据包丢失带来的影响，从而改善视频回放质量。     

基于GA-BP的实时视频通信自适应前向纠错码

针对在动态的网络环境中,实时视频在基于数据报协议的传输中遇到大量丢包的问题,提出了一种基于遗传算法改进的BP神经网络自适应前向纠错码的方法。该方法通过设计自适应的前向纠错编码来恢复丢失的数据包。首先,设计一个基于帧级别的前向纠错的框架,主要包括视频编码解码器、RS前向纠错码编解码模块和前向纠错码冗余度计算模块,用来模拟一般视频传输的环境;然后,设计GA-BP模型,将其应用到RS-FEC的冗余度计算,实现一种帧级不均等的保护方案;最后,基于GE信道模拟网络,生成丢包数据,训练离线模型,进行实验验证。实验结果表明,相比StaticRS和DeepRS,所提方法能够在较低冗余度下较高地恢复丢失的数据包,得到更高质量的视频。     

基于位置重排的视频不等纠错保护方法

针对基于DCT变换的视频压缩码流,该文提出了一种可有效改善其抗误码性能的收发交互不等纠错保护算法．主要思想是在宏块层对不同数据信息进行位置重排,使DCT系数按重要性重新排列,然后通过EREC算法提高宏块头等同步信息的抗误码性能．在信道编码部分,该算法依据译码端反馈的信道状态参数实时调整待传输的比特数和RCPC编码速率,从而实现收发交互的不等纠错保护(IUEP)．仿真结果表明,该文算法在高误码率信道下能较大程度地改善解码PSNR值,其性能明显优于UEP、EEP等方法．     

基于错误传播模型的非均等视频流丢失保护

提出一种适用于丢包网络、面向图像组（GOP）层的非均等视频流丢失保护方案。利用GOP中不同帧之间的非均等显著性,将不同数量前向错误校正包分配到GOP层的不同帧中。采用帧间包交错机制将突发包丢失分散到不同帧上,提高处理突发包丢失时的鲁棒性。仿真结果表明,在不同信道丢失模式下,该方案能提高视频接收质量。     

Concurrent multipath transmission combining forward error correction and path interleaving for video streaming

Video streaming is a popular application on next generation networks (NGNs). However, video streaming over NGNs has many challenges due to the high bit error rates of these networks. Forward error correction (FEC) is often applied to improve the quality of video streaming. However, continuous lost packets decrease the recovery performance of FEC protection in NGNs. To disperse continuous lost packets to different FEC blocks, we propose a concurrent multipath transmission that combines FEC with path interleaving. Our proposed control scheme adaptively adjusts the FEC block length and concurrently sends data interleaved over multiple paths. Experimental results with our approach show improved packet loss and signal to noise ratio performance.     

A low complexity model for predicting slice loss distortion for prioritizing H.264/AVC video

The cumulative mean squared error (CMSE) is a widely used measure of distortion introduced by a slice loss. We propose a low-complexity and low-delay generalized linear model for predicting CMSE contributed by the loss of individual H.264/AVC encoded video slices. We train the model over a video database by using a combination of video factors that are extracted during the encoding of the current frame, without using any data from future frames in the group of pictures (GOP). We then analyze the accuracy of the CMSE prediction model using cross-validation and correlation coefficients. We prioritize the slices within a GOP based on their predicted CMSE values. The performance of our model is evaluated by applying unequal error protection, using rate compatible punctured convolutional codes, to the prioritized slices over noisy channels. We also demonstrate an application of our slice prioritization by implementing a slice discard scheme, where the slices are dropped from the router when the network experiences congestion. The simulation results show that (i) the slice CMSE prediction model performs well for varying GOP structures, GOP lengths, and encoding bit rates, and (ii) the peak signal-to-noise ratio and video quality metric performance of an unequal error protection algorithm using slices prioritized by the predicted CMSE is similar to that of the measured CMSE values for different videos and channel signal-to-noise. We also extend the GOP-level slice prioritization to frame-level slice prioritization and show its performance over noisy channels.     

基于IP无线网络FGS视频传输的多乘积码方案

研究基于IP无线网络中精细粒度可伸缩性(FGS)视频的传输。基于包交换的IP无线网络通常由两段链路组成:有线链路和无线链路。为了处理这种混合网络中不同类型数据包的丢失情况,对FGS视频增强层数据运用了一个具有比特平面间不平等差错保护(BPUEP)的多乘积码前向纠错(MPFEC)方案进行信道编码。对FGS增强层每一个比特平面(BP),在传输层,采用里德—索罗蒙码(RS)提供比特平面间的保护;而在链路层,则运用循环冗余校验码(CRC)串联率兼容穿孔卷积码(RCPC)提供数据包内保护。还提出了一个率失真优化的信源—信道联合编码的码率配置方案,仿真结果显示出该方案在提高接收端视频质量方面的优势。     

一种面向视频应用的有效实时码率控制算法

为提供实时视频传输服务,同时保证一定的视频质量,提出了一种基于JVT-G012的动态码率控制算法。在编码过程中,根据图像内容的运动特性,动态决定GOP长度,同时防止输出缓冲器的上溢和下溢;为复杂度较高的宏块（MB）分配较多的比特。实验结果表明,该算法对于场景变化较大的实际视频序列有较好的编码效果。     

An unequal packet loss resilience scheme for video over the Internet

We present an unequal packet loss resilience scheme for robust transmission of video over the Internet. By jointly exploiting the unequal importance existing in different levels of syntax hierarchy in video coding schemes, GOP-level and Resynchronization-packet-level Integrated Protection (GRIP) is designed for joint unequal loss protection (ULP) in these two levels using forward error correction (FEC) across packets. Two algorithms are developed to achieve efficient FEC assignment for the proposed GRIP framework: a model-based FEC assignment algorithm and a heuristic FEC assignment algorithm. The model-based FEC assignment algorithm is to achieve optimal allocation of FEC codes based on a simple but effective performance metric, namely distortion-weighted expected length of error propagation, which is adopted to quantify the temporal propagation effect of packet loss on video quality degradation. The heuristic FEC assignment algorithm aims at providing a much simpler yet effective FEC assignment with little computational complexity. The proposed GRIP together with any of the two developed FEC assignment algorithms demonstrates strong robustness against burst packet losses with adaptation to different channel status.     

Adaptive hybrid error correction model for video streaming over wireless networks

The Hybrid ARQ (HARQ) mechanism is the well-known error packet recovery solution composed of the Automation Repeat reQuest (ARQ) mechanism and the Forward Error Correction (FEC) mechanism. However, the HARQ mechanism neither retransmits the packet to the receiver in time when the packet cannot be recovered by the FEC scheme nor dynamically adjusts the number of FEC redundant packets according to network conditions. In this paper, the Adaptive Hybrid Error Correction Model (AHECM) is proposed to improve the HARQ mechanism. The AHECM can limit the packet retransmission delay to the most tolerable end-to-end delay. Besides, the AHECM can find the appropriate FEC parameter to avoid network congestion and reduce the number of FEC redundant packets by predicting the effective packet loss rate. Meanwhile, when the end-to-end delay requirement can be met, the AHECM will only retransmit the necessary number of redundant FEC packets to receiver in comparison with legacy HARQ mechanisms. Furthermore, the AHECM can use an Unequal Error Protection to protect important multimedia frames against channel errors of wireless networks. Besides, the AHECM uses the Markov model to estimate the burst bit error condition over wireless networks. The AHECM is evaluated by several metrics such as the effective packet loss rate, the error recovery efficiency, the decodable frame rate, and the peak signal to noise ratio to verify the efficiency in delivering video streaming over wireless networks.     

An error-resilient GOP structure for robust video transmission

Recent advances in technology have resulted in a significant growth in wireless communications and widespread access to information via the Internet, which have resulted in a strong demand for reliable transmission of video data. The challenge of robust video transmission is to protect the compressed data against hostile channel conditions while bringing little impact on bandwidth efficiency. In motion-compensated video-coding schemes, such as MPEG-1 or MPEG-2, an I frame normally is followed by several P frames and possibly B frames in a group-of-pictures (GOP). In error-prone environments, error happening in the previous frames in a GOP may propagate to all the following frames until the next I frame, which is the beginning of the next GOP. In this paper, we propose a novel GOP structure for robust transmission of MPEG video bitstream. By selecting the optimal position of the I frame in a GOP, robustness can be achieved without reducing any coding efficiency. Another advantage of the proposed GOP structure is also analyzed: compared with the conventional GOP structure, it provides reverse-play operation for MPEG video streaming with much less requirement on the network bandwidth. Experimental results demonstrate both the robustness of the proposed GOP structure and the efficient reverse-play functionality it leads to.     

适合网络传输的鲁棒小波低比特率视频编码

提出了一种适合包交换网络传输的基于离散小波变换的视频编码方案,通过对SPIHT小波系数编码算法进行复杂度降低、纹理分割等修改,来适应视频编码在编码效率和鲁棒性方面的要求,为解决网络数据包丢失造成的帧质量骤降,方案对数据包的重要性进行了均衡,每一个数据包中均包含帧内信息和帧间信息,利用改进的SPIHT算法生成混合比特流。试验表明,该方案运算复杂度低,对网络传输中的包丢失不敏感,并且能很好地抑制错误传播。     

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。     

低延时无线视频传输中的H.264码率控制算法

针对基于ARQ机制下的低延时无线视频传输,提出了一种有效的信道信源联合码率控制方案。该方案首先利用Markov模型作为无线信道模型来预测估计信道的状态和带宽,并以基于Cauchy分布的率失真模型作为信源模型;然后在帧层设计了一种基于PID控制器的缓存控制算法,即根据预测信道带宽和缓存器状态来分配每帧目标比特数,以提高缓存的控制能力;对于宏块层则在监控缓存器状态的同时,利用基于Cauchy分布的率失真模型,最后通过Lagrange方法来优化分配目标比特数。实验结果证明,该算法不仅能显著提高平均峰值信噪比,并能大幅减少跳帧数目。     

Robust transmission of scalable video stream using modified LT codes

LT codes are convenient and popular kind of rateless codes that could easily tolerate different patterns of loss in erasure channels. In this paper an LT code with unequal packet protection (UPP) property is proposed. The proposed code could provide unequal packet recovery to any importance-sorted data packets. Simulation results indicate the enhanced performance of the suggested scheme and its ability to increase the probability of early decoding of more important parts of data rather than the rest. Also it is shown that the proposed scheme could provide comparable bit error rates in comparison to the one of the well known previous methods. The code with modified encoding graph has been utilized for transmitting of the scalable data-partitioned video stream. Simulation results also illustrate the performance of the suggested approach in early delivery of the most important parts of a video sequence.     

Classification of slice-based VBR video traffic and estimation of link loss by exceedance

Classification of a video stream is an essential preliminary step to estimate the bit loss when the video stream is transmitted over a communication network. In this paper, we classify the video frames by the average frame size and estimate the bit loss for each class when the bitrate exceeds the capacity of the bottleneck link. The video stream under study is encoded using the explicit slice-based H.264/AVC encoding scheme. This scheme reduces the burstiness of regular H.264/AVC encoded video by removing the traditional GOP structure. Instead, a repetitive combination of intracoded and predicted slices is employed, thereby introducing a specific dependence structure in the video data. We consider a bufferless model of the communication system and evaluate the channel capacity required to give a maximum allowed loss rate for each class.Due to the high variability, non-stationarity and non-homogeneity of the underlying video data, the obtained classes are checked regarding the dependence and distribution structure of the data. The high quantiles of the losses are estimated for each class.     

Adaptive residual-based distributed compressed sensing for soft video multicasting over wireless networks

Recently, multicasting of video signals has become a useful technology in wireless networks, in which the main challenge is to scalably serve multiple receivers that have different channel characteristics. In this paper, we propose an adaptive residual-based distributed compressed-sensing scheme for soft video multicast (ARDCS-cast). At the encoder, we first adaptively determine if a block in a non-reference frame should be measured directly or predictively during compressed-sensing. The resulting adaptive measurements from non-reference frames are then packeted together with the measurements of the reference frames. We further derive the optimal power allocation scheme for the measurements from each frame within each packet. The packets are then transmitted over the wireless channel. At the decoder, the receivers with different channel characteristics obtain different numbers of packets and reconstruct videos with different quality. Experimental results show that the proposed ARDCS-cast is more effective than the state-of-the-art SoftCast-2D, SoftCast-3D and DCS-cast schemes in both unicast and multicast scenarios.     

Packet Permutation: A Robust Transmission Technique for Continuous Media Streaming Over the Internet

With the growing popularity of the Internet, there is an increasing demand to deliver continuous media (CM) streams over the Internet. However, packets may be damaged or lost during transmission over the current Internet. In particular, periodic network overloads often result in bursty packet losses, degrading the perceptual quality of CM streaming. In this paper, we focus on reducing the impact of this bursty loss behavior. We propose a novel robust end-to-end transmission scheme, referred to as packet permutation (PP), to deliver pre-compressed continuous media streams over the Internet. At the server side, PP permutes, prior to transmission, the normal packet delivery sequence of CM streams in a specific way. The packets are then re-permuted at the receiver side before they are presented to the application. In this way, the probability of losing a large number of packets within each CM frame can be significantly reduced. To validate the effectiveness of PP, a series of trace-driven simulations are conducted. Our results show that for a given quality of service (QoS) requirement of CM streaming, PP greatly reduces the overhead required by traditional error control schemes, such as forward error correction (FEC) and feedback/retransmission-based schemes.