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.     

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.     

基于集成FEC和层次传输的可靠组播的流控技术

端系统能力和网络带宽的异构性给大规模组播的流控带来很大困难。本文将前向纠错（ＦＥＣ）技术与层次传输相结合,较好地解决了异构环境下可靠组播的流控问题。我们给出了传输调度和信道速率分配的算法,讨论了差错控制问题。性能分析和模拟表明,该方法对大规模、异构组播组可显著减少平均传输时间并且有效地利用网络带宽。只需较少数目的组播组就能得到性能的很大提高。软件ＦＥＣ编码器的速度能够匹配当前的网络条件,算法易于实     

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 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.     

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。     

Relay-assisted hierarchical adaptation scheme for multi-user scalable video delivery to heterogeneous mobile devices

In multi-user video (MUV) delivery scenarios,the available resources of receiver devices,such as processing capability,link packet error rate (PER),and bandwidth,are usually different.We propose a relay-assisted hierarchical adaptation (RHA) scheme to maximize the total perceptual quality of all users when transmitting video streams coded via scalable video coding (SVC).First,MUV bitstreams are adaptively extracted under the constraints of network bandwidth and individual decoding capacity.Next,the relay links are introduced as substitutes of possible bad direct links for packets retransmissions.Approximately equal opportunity of transmission is allocated to each stream while the packets inside a stream are scheduled according to their priorities.The priorities are determined by the links states and packets loss distortions.Simulation results show that our RHA scheme has significant performance improvements compared with other schemes.     

Header Detection to Improve Multimedia Quality Over Wireless Networks

Wireless multimedia studies have revealed that forward error correction (FEC) on corrupted packets yields better bandwidth utilization and lower delay than retransmissions. To facilitate FEC-based recovery, corrupted packets should not be dropped so that maximum number of packets is relayed to a wireless receiver's FEC decoder. Previous studies proposed to mitigate wireless packet drops by a partial checksum that ignored payload errors. Such schemes require modifications to both transmitters and receivers, and incur packet-losses due to header errors. In this paper, we introduce a receiver-based scheme which uses the history of active multimedia sessions to detect transmitted values of corrupted packet headers, thereby improving wireless multimedia throughput. Header detection is posed as the decision-theoretic problem of multihypothesis detection of known parameters in noise. Performance of the proposed scheme is evaluated using trace-driven video simulations on an 802.11b local area network. We show that header detection with application layer FEC provides significant throughput and video quality improvements over the conventional UDP/IP/802.11 protocol stack     

一种自适应的视频流化前向纠错算法

网络视频应用经常会受到数据包丢失或错误以及网络带宽资源不足的干扰.相关研究表明:在多数情况下,动态变化的网络带宽和丢包率是影响视频流化质量的关键因素.因此,为了保证视频质量,可以采用前向纠错(forward error correction,简称FEC)编码来提高视频数据传输的可靠性;同时,为了适应网络状态的变化,发送端可以调节视频数据的发送速率,并在视频源数据与FEC数据之间合理分配网络传输带宽.首先通过对视频流结构的分析,在充分考虑帧之间的依赖关系和帧类型的基础上提出了一种帧的解码模型.在此基础上,建立了用于在视频源数据和FEC数据之间分配网络带宽资源的优化算法.实验表明,该模型可以有效地适应网络状态的变化,并通过优化分配网络带宽资源来使接收端获得最大的可播放帧率.     

Generic forward error correction of short frames for IP streaming applications

If the frame size of a multimedia encoder is small, Internet Protocol (IP) streaming applications need to pack many encoded media frames in each Real-time Transport Protocol (RTP) packet to avoid unnecessary header overhead. The generic forward error correction (FEC) mechanisms proposed in the literature for RTP transmission do not perform optimally in terms of stability when the RTP payload consists of several individual data elements of equal priority. In this paper, we present a novel approach for generating FEC packets optimized for applications packing multiple individually decodable media frames in each RTP payload. In the proposed method, a set of frames and its corresponding FEC data are spread among multiple packets so that the experienced frame loss rate does not vary greatly under different packet loss patterns. We verify the performance improvement gained against traditional generic FEC by analyzing and comparing the variance of the residual frame loss rate in the proposed packetization scheme and in the baseline generic FEC.     

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。     

Video compression using MPEG 7-MBBMC: techniques for communication

Communication is termed as exchanging the information (audio, video, text and image) from one end (transmitter) to another end (receiver). When video data are compressed and transmitted to another side, compression reduces the bandwidth size and memory required to transmit the video. Some traditional techniques are used in video transmission but it includes drawbacks, such as more compression time and low quality due to compression. To overcome these drawbacks the MPEG7-MBBMC (Modified Block Based Motion Compensated) technique is developed. Here the input video signals are collected from the dataset and the signals are splitted into three bands. Discrete Wavelet Transform (DWT) is applied for each band and quantization process occurs. The DWT and quantization process are applied in the MPEG7 compression, which offers high compression factors. Next, encoder is applied to convert the packets into small packets by using modified block based motion compensated (MBBMC) technique. The Motion compensation establishes a correspondence between elements of nearby images in the video sequence. The Forward Error Correction (FEC) is used to reduce the distortion in the encoder video packet. Then the Channel Pattern Integration (CPI) is applied to find the best available channel. The encoded video packets are transmitted by the best available channel. In receiver side the error correction code is applied to decode the video packets and reconstructs the decoded packet by decompression. It improves the quality of the video and in future it will help for much development in the field of multimedia.

     

一种联合速率控制和缓冲管理的VBR视频鲁棒传输方法

In this paper we present an adaptive video transmission framework that integrates rate allocation and buffer control at the source with the playback adjustment mechanism at the receiver. A transmission rate is determined by a rate allocation algorithm which uses the program clock reference (PCR) embedded in the video streams to regulate the transmission rate in a refined way. The server side also maintains multiple buffers for packets of different importance levels to trade off random loss for controlled loss according to the source buffer size, the visual impact, and the playback deadline. An over-boundary playback adjustment mechanism based on proportional-integra (PI) controller is adopted at the receiver to maximize the visual quality of the displayed video according to the overall loss and the receiver buffer occupancy. The performance of our proposed framework is evaluated in terms of peak signal-to-noise ratio (PSNR) in the simulations, and the simulation results demonstrate the improvement of the average PSNR values as well as the better quality of the decoded frames.     

Packet loss resilience using unequal forward error correction assignment for video transmission over communication networks

Forward error correction (FEC) methods have been developed for packet loss resilience in application layer for real-time video transmission over communication networks. In this paper, an efficient packet loss resilience method is proposed using closed form solution for unequal FEC assignment based on a new packet distortion model. We first derive the packet distortion model by investigating the error concealment property and error propagation effect in H.264. To select the source and channel rate minimizing the overall distortion, we present a model-based rate allocation algorithm using the packet distortion model and rate-distortion function. Then we propose the closed form solution for unequal FEC assignment, which uses the packet distortion model and considers channel status information. Simulation results show that the proposed method gives substantial improvement for the received video quality in packet-lossy Internet and wireless network environments, while it requires much less computational complexity compared to the previous scheme.     

Adaptive robust video broadcast via satellite

With increasing demand for multimedia content over channels with limited bandwidth and heavy packet losses, higher coding efficiency and stronger error resiliency is required more than ever before. Both the coding efficiency and error resiliency are two opposing processes that require appropriate balancing. On the source encoding side the video encoder H.264/AVC can provide higher compression with strong error resiliency, while on the channel error correction coding side the raptor code has proven its effectiveness, with only modest overhead required for the recovery of lost data. This paper compares the efficiency and overhead of both the raptor codes and the error resiliency techniques of video standards so that both can be balanced for better compression and quality. The result is also improved by confining the robust stream to the period of poor channel conditions by adaptively switching between the video streams using switching frames introduced in H.264/AVC. In this case the video stream is initially transmitted without error resiliency assuming the channel to be completely error free, and then the robustness is increased based on the channel conditions and/or user demand. The results showed that although switching can increase the peak signal to noise ratio in the presence of losses but at the same time its excessive repetition can be irritating to the viewers. Therefore to evaluate the perceptual quality of the video streams and to find the optimum number of switching during a session, these streams were scored by different viewers for quality of enhancement. The results of the proposed scheme show an increase of 3 to 4 dB in peak signal to noise ratio with acceptable quality of enhancement.     

Scalable Video Multicast Using Expanding Window Fountain Codes

Fountain codes were introduced as an efficient and universal forward error correction (FEC) solution for data multicast over lossy packet networks. They have recently been proposed for large scale multimedia content delivery in practical multimedia distribution systems. However, standard fountain codes, such as LT or Raptor codes, are not designed to meet unequal error protection (UEP) requirements typical in real-time scalable video multicast applications. In this paper, we propose recently introduced UEP expanding window fountain (EWF) codes as a flexible and efficient solution for real-time scalable video multicast. We demonstrate that the design flexibility and UEP performance make EWF codes ideally suited for this scenario, i.e., EWF codes offer a number of design parameters to be ldquotunedrdquo at the server side to meet the different reception criteria of heterogeneous receivers. The performance analysis using both analytical results and simulation experiments of H.264 scalable video coding (SVC) multicast to heterogeneous receiver classes confirms the flexibility and efficiency of the proposed EWF-based FEC solution.     

Combining Media-Specific FEC and Error Concealment for Robust Distributed Speech Recognition Over Loss-Prone Packet Channels

This paper presents a mixed recovery scheme for robust distributed speech recognition (DSR) implemented over a packet channel which suffers packet losses. The scheme combines media-specific forward error correction (FEC) and error concealment (EC). Media-specific FEC is applied at the client side, where FEC bits representing strongly quantized versions of the speech vectors are introduced. At the server side, the information provided by those FEC bits is used by the EC algorithm to improve the recognition performance. We investigate the adaptation of two different EC techniques, namely minimum mean square error (MMSE) estimation, which operates at the decoding stage, and weighted Viterbi recognition (WVR), where EC is applied at the recognition stage, in order to be used along with FEC. The experimental results show that a significant increase in recognition accuracy can be obtained with very little bandwidth increase, which may be null in practice, and a limited increase in latency, which in any case is not so critical for an application such as DSR     

基于二项式概率的冗余传输策略

分析现有音视频传输的丢包恢复技术,结合前向纠错Reed-Solomon冗余编码及交织恢复丢包技术提出二项式概率模型。该模型根据接收端反馈的结果计算需要编码的冗余包个数,使用交织技术将音频冗余包与原始数据混合传输,有效节省了带宽资源。实验结果表明,该模型在网络拥塞的情况下,能根据实际情况产生足够的冗余数据包,使接收端收到数据后还原出原始数据并播放,提高了音视频的传输质量和播放质量。     

Temporally enhanced erasure codes for reliable communication protocols

In forward error correction (FEC), redundant encoded packets are transmitted to and decoded by receiver(s) so that up to a certain number of lost packets can be recovered by using those redundant packets. The data stream is divided into blocks, and encoded packets are generated within each block. Hence, the encoding and decoding are performed on block-by-block basis.In this paper, a new mechanism named as temporally enhanced FEC (TEFEC) is proposed as an enhancement to the existing block-based FEC codes such as erasure codes. In TEFEC, the scopes of encoding and decoding are expanded beyond block boundaries, and they may overlap with scopes of neighboring blocks. To show its feasibility, TEFEC is applied to erasure codes to enhance their error correction capability, and new codes named as temporally enhanced erasure codes (TEEC) are developed and presented in this paper. In addition, to show its advantages, a reliable protocol was designed and simulated by combining both automatic repeat request and TEEC techniques. In terms of average packet retransmission rates and end-to-end delays for packet delivery, it is shown that the protocol employing TEEC outperforms block-by-block based protocols in most of the cases.     

一种MIMO-HARQ无线系统的接收机设计

混合自动请求重传(HARQ)以ARQ方式合并带有前向纠错码(FEC)的数据包,明显地提高系统的吞吐量。围绕无线通信系统中几种MIMO-HARQ合并方案,从存储要求、误码率等多角度考虑接收机的设计。提出了一种新的接收机合并方案,分析和仿真表明,所提方案具有很好的误码率性能与存储要求之间的折中,可应用于Chase合并和增量冗余合并中。