Rate-smoothed schedule with tolerable data dropping for video coding stream

Most algorithms of smoothing schedule compute the required bit rate of video transmission to satisfy all the transmitted data. In this paper, our proposed tolerable data dropping algorithm can adjust transmitting data to fit available bit rate. MPEG-4 with fine grained scalability (FGS) can support partial data dropping to adapt to available bandwidth network. The algorithm is based on the minimum variance bandwidth allocation (MVBA) algorithm proposed by Salehi et al. to compute the bit rate such that still ensuring that the buffer never underflows and overflows for MPEG-4 FGS streams under the limited bandwidth resource. We prove that our proposed algorithm, named MVBADP, is smoother than the MVBA algorithm. The experimental results show the peak rate, the number of rate changes, and the ratio of total dropping data, and the PSNR for four test sequences with different content characteristics. They are varied by buffer sizes and tolerable dropping ratios. We found that the MVBADP algorithm can reduce the peak rate and the number of changes when the transmitted data are dropped by tolerable dropping ratio, especially on the video sequences with the high motion and complex texture characteristic and larger size change of the consecutive frame.     

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.     

Online smoothing of VBR H.263 video for the CDMA2000 and IS-95B uplinks

The variable bit rate nature of compressed video remains a major challenge to the transmission of real-time video over mobile CDMA cellular networks. Indeed, H.263 and MPEG-4 video bitstreams can exhibit high peak rates and frequent rate variations, which are difficult to support in 2.5G and 3G mobile networks. In this paper, we consider smoothing schemes to minimize the peak rate, variance, and average bandwidth of the transmitted signal, in order to obtain a lower bit-error-rate and thus, a higher received video quality than if unsmoothed video were transmitted. By introducing a delay in the decoding process, the video output stream can be buffered at the video decoder end: this buffering capability is used in order to perform real-time smoothing of the video streams. The transmitter chooses among the set of rates provided by the IS-95B or cdma2000 standards, in a mobility scenario, in order to support the rate variations. We simulate two end-to-end video communication systems based on IS-95B and cdma2000. They consist of an H.263 codec, an H.223 multiplexer, a rate management system and the physical layer components of the IS-95B and cdma2000 standards. The transmitted signal is subject to multipath fading and multiple-access interference, obtained by simulating all the other users in the cell. Results show a significant gain in the video quality obtained through smoothing of the video streams.     

基于漏斗的实时VBR视频最短路径平滑算法①

针对实时VBR视频流式传输的在线平滑优化问题,提出一种基于漏斗的最短路径平滑算法——SPSF。SPSF利用滑动窗口对实时VBR视频进行分段处理,顺序读取和缓存每帧视频数据至窗口,并基于漏斗原理求解窗口内数据的最短路径。数据填满窗口后根据求得的最短路径进行传输,同时根据路径特征推进窗口滑动进行下一段数据的平滑处理及传输,以此类推完成整个视频平滑传输。实验结果表明。与传统的在线平滑算法相比,SPSF具有更优的传输比特率峰值、传输比特率谷值、及传输比特率方差;与传统的最短路径算法相比,SPSF具有更快的最短路径求解速度,提高了视频传输的实时性。     

基于漏斗的实时VBR视频最短路径平滑算法①

针对实时VBR视频流式传输的在线平滑优化问题,提出一种基于漏斗的最短路径平滑算法——SPSF。SPSF利用滑动窗口对实时VBR视频进行分段处理,顺序读取和缓存每帧视频数据至窗口,并基于漏斗原理求解窗口内数据的最短路径。数据填满窗口后根据求得的最短路径进行传输,同时根据路径特征推进窗口滑动进行下一段数据的平滑处理及传输,以此类推完成整个视频平滑传输。实验结果表明,与传统的在线平滑算法相比,SPSF具有更优的传输比特率峰值、传输比特率谷值、及传输比特率方差;与传统的最短路径算法相比,SPSF具有更快的最短路径     

基于H.264运动矢量域的脆弱水印算法研究

针对H.264/AVC的编码特征,提出一种基于H.264/AVC低比特率视频流的脆弱水印算法。该算法在视频编码端的P帧经过运动矢量搜索和运动矢量预测之后,提取运动矢量残差,将水印嵌入在运动矢量残差中。在视频的解码端进行水印提取和检测,检测不需要原始视频的参与。该算法具有较小的码率变化、视频失真,以及较强的脆弱性。     

结合帧率变换与HEVC标准的新型视频压缩编码算法

相比于之前主流的H.264视频压缩编码标准,HEVC在保证重建视频质量相同的前提下,可以将码率降低近50%,节省了传输所需的带宽.即便如此,由于一些特定的网络带宽限制,为继续改善HEVC视频编码性能,进一步提升对视频的压缩效率仍然是当前研究的热点.本文提出一种HEVC标准编码与帧率变换方法相结合的新型的视频压缩编码算法,首先在编码端,提出一种自适应抽帧方法,降低原视频帧率,减少所需传输数据量,对低帧率视频进行编解码;在解码端,结合从HEVC传输码流中提取的运动信息以及针对HEVC编码特定的视频帧的分块模式信息等,对丢失帧运动信息进行估计;最后,通过本文提出的改进基于块覆盖双向运动补偿插帧方法对视频进行恢复重建.实验结果证实了本文所提算法的有效性.     

Effective macroblock layer rate control algorithm for H.264/AVC

Rate control plays a critical role in the video encoder. In this paper, an effective macroblock (MB) layer rate control algorithm is proposed for H.264/AVC to improve video quality and reduce the computational complexity. By thoroughly analyzing the temporal–spatial correlation and object direction, the mean absolute difference (MAD) value of current MB is a weighted combination of temporal MAD predicted by previous frame and spatial MAD predicted by current frame, and then four directional patterns are chosen adaptively for data point selection in quadratic rate-quantization (R-Q) model parameters estimation. Finally, a more accurate header bits prediction model is developed to improve the accuracy of allocated bits which can be used for quantization parameter (QP) calculation. Extensive experiment results show that compared with JVT-G012, the proposed algorithm achieves higher average peak signal to noise ratio (PSNR) up to 0.51 dB improvements and more accurate target bit rates.     

端到端MPEG-4 FGS视频TCP友好的平滑传输

着重研究了Internet上MPEG-4 FGS(fine grained scalable)视频流的自适应平滑传输,其主要目的在于,在网络带宽变化的情况下,提供稳定的视频回放质量.提出了一种新的基于TFRC(TCP-friendly rate control)的MPEG-4 FGS端到端视频流传输系统框架,在此框架的基础上,首先假设完整的可用带宽变化已知,并且提出了一种离线的自适应平滑算法.此后,给出一种基于改进的ARAR(autoregressive autoregressive)预测技术的在线自适应平滑算法.最后,以NS-2为实验平台进行了模拟实验.模拟实验表明,提出的离线和在线自适应平滑算法可以充分利用可用网络带宽,并且能够在可用网络带宽持续波动的情况下保证接收方的回放尽可能地平稳,从而达到获得最佳视觉效果的目的.     

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.     

Fast macroblock mode decision for H.264/AVC baseline profile video transcoder based on support vector machines

In video transcoding, accuracy and efficiency of macroblock mode decision are critical issues at the re-encoder side due to the changes in frame size, frame rate, and bit rate. In this paper, a fast macroblock mode decision scheme based on support vector machines is proposed for H.264/AVC baseline profile video transcoder. Features including motion vectors, residual data, pre-encoded macroblock modes, and quantization parameters are extracted from incoming bitstream in both of training stage and classification stage. Feature extraction methods are investigated for spatial resolution transcoder, temporal resolution transcoder, and bit-rate transcoder. After off-line training and simplification of support vectors, the obtained support vector machine classifier can determine macroblock mode in the re-encoder accurately. Extensive experiments are carried out on different types of transcoders and results show that the proposed method can save about 80% in computational complexity compared to full mode search algorithm implemented in the latest H.264/AVC reference software (JM17.1), while maximum peak signal-to-noise ratio is degraded by 0.2–1.1?dB depending on different sequences and bit rate.     

基于运动矢量细化的帧率上变换与HEVC结合的视频压缩算法

将帧率变换技术与新型视频压缩编码标准HEVC相结合有利于提升视频的压缩效率。针对直接利用HEVC码流信息中的低帧率视频的运动矢量进行帧率上变换时效果不理想的问题,文中提出了一种基于运动矢量细化的帧率上变换与HEVC结合的视频压缩算法。首先,在编码端对原始视频进行抽帧,降低视频帧率;其次,对低帧率视频进行HEVC编解码;然后,在解码端与从HEVC码流中提取出的运动矢量相结合,利用前向-后向联合运动估计对其进行进一步的细化,使细化后的运动矢量更加接近于对象的真实运动;最后,利用基于运动补偿的帧率上变换技术将视频序列恢复至原始帧率。实验结果表明,与HEVC标准相比,所提算法在同等视频质量下可节省一定的码率。同时,与其他算法相比,在节省码率相同的情况下,所提算法重建视频的PSNR值平均可提升0.5 dB。     

EPCRTT-based smoothing and multiplexing of VBR video traffic

Applying video smoothing techniques to real-time video transmission can significantly reduce the peak rate and rate variability of compressed video streams. Moreover, statistical multiplexing of the smoothed traffic can substantially improve network utilization. In this paper we propose a new smoothing scheme, which exploits statistical multiplexing gain that can be obtained after smoothing of individual video streams. We present a new bandwidth allocation algorithm that allows for responsive interactivity. The local re-smoothing algorithm is carried out using an iterative process. In the proposed scheme the smoothed video streams are divided into fixed intervals and then a new transmission schedule for each interval is calculated. The problem of applying an optimal transmission schedule for aggregated smoothing video streams is shown to be NP-hard problem. Partitioning the whole stream into sections enables parallel processing of the smoothing algorithm in real-time before transmission. This approach allows partial transmission of the multiplexed stream while smoothing other intervals. The simulation results show a significant reduction in peak rate and rate variability of the aggregated stream, compared to the non-smoothing case. Therefore the proposed scheme allows us to increase the number of simultanusally-served video streams.

Real-time traffic transmission over the Internet

Multimedia applications require the transmission of real-time streams over a network. These streams often exhibit variable bandwidth requirements, and require high bandwidths and guarantees from the network. This creates problems when such streams are delivered over the Internet. To solve these problems, recently, a small set of differentiated services has been introduced. Among these, Premium Service is suitable for transmitting real-time stored stream (full knowledge of the stream characteristics). It uses a bandwidth allocation mechanism (BAM) based on the stream peak rate. Due to the variable bandwidth requirement, the peak rate BAM can waste large amount of bandwidth. In this paper we propose a new BAM that uses less bandwidth than the peak rate BAM, while providing the same service. Our BAM does not affect the real-time stream quality of service (QoS) and does not require any modification to the Premium Service Architecture. We also introduce several frame dropping mechanisms that further reduce bandwidth consumption subject to a QoS constraint when coupled with the above BAM. The proposed BAM and the dropping mechanisms are evaluated using Motion JPEG and MPEG videos and are shown to be effective in reducing bandwidth requirements. Further, since VCR operations are very useful in video streaming, we propose a mechanism that introduces these operations in our BAM. Through simulations we show the effectiveness of this mechanism     

Wireless transmission of HD video using H.264 compression over UWB channel

With the technological advancement, entertainment has become revolutionized and the high-definition (HD) video has become an integral feature of our modern amusement system. The demand for wireless transmission of HD video is rapidly rising for its ubiquitous nature, easy installation and relocation. Such wireless transmissions of HD video streams require very high bandwidth. The ultra-wideband (UWB) offers a large bandwidth, and short-range high-speed data transmission at low cost and low power consumption. In this paper, we present the feasibility study to transmit HD video wirelessly using H.264/AVC compression over the UWB communication channel. Simulations are carried out by controlling key H.264/AVC encoder parameters such as, in-loop deblocking filter, group of pictures, and quantization parameter. Based on the analysis, an optimum setting of these parameters is proposed for different bandwidth requirements, as well as acceptable video quality. The bandwidth achieved is restricted between 1.5 and 20?Mbps with a minimum reconstruction quality of 34?dB. The HD bit stream is then transmitted over the UWB communication channel and the demonstration shows that the overall encoder performance is satisfactory with the transmission bit-error-rate (BER) in the range of 10^?5?C10^?8.     

Content-based adaptive compression of educational videos using phase correlation techniques

A serious bottleneck towards multimedia e-learning is the non-availability of required bandwidth to view the lecture videos at good resolution because of large size of lecture videos. Content-based compression of video data can greatly enhance the bandwidth utilization over scarce resource networks. In this paper, an educational video compression technique is presented that dynamically allocates the space according to the content importance of each video segment in the educational videos. We present a phase-correlation-based motion estimation and compensation algorithm that assists in the compression of important moving objects in an efficient manner. Temporal coherence is exploited in a two-phase manner. First, the frames with high similarity are categorized and encoded efficiently. Second, the compression ratio is adapted according to the frame content. Shots that are of high importance are stored at a higher bit rate as compared to the frames of relatively low importance. The importance and priority of the frames is computed automatically by our algorithm. Results over several hours of educational videos and comparison with the state-of-the-art compression algorithms illustrate the high compression performance of our technique.     

异构环境下层次编码多视频源多共享信道分层组播

视频组播是许多当前和将来网络服务的重要组成部分,如视频会议,远程学习、远地展示及视频点播,随着网络传送基础设施的改善和端系统处理能力的增强,组播视频应用日益变得可行,组播视频传输中存在的主要问题是网络送资源的异构性和动态性,其使得视频流的多个接收方都达到可接受的流量特性变得异常困难,目前该问题的一个有效解决方式就是利用自适应的分层视频传输机制,在该机制中,各源产生层次媒体流,并在多个网络信道中传输。对视频会议类的多点到多点视频组播应用,信道往往被所有潜在的发送方共享,任何发送方都可在任何一个共享信道中发送其视频层次。在该多点到多点、共享信道、分层视频组播模型下,一个关键问题就是如何动态确定各视频源层次到各共享组播信道的映射,映射策略直接影响到会话整体视频接收质量和网络带宽利用率。典型的方式是顺序映射,该映射方式同等对待各发送方,但利用该方式,随源数目的增加,在各共享网络信度上会出现带宽可伸缩性问题,而且顺序映射方式无法适应网络传送资源和会话状态的动态变化。为此,该文设计了一种基于接收方反馈信息的自适应的层次映射算法,接收方周期性地将其当前感兴趣的发送方及接收速率的信息反馈给某控制节点,而控制节点就利用当前反馈信息动态地调整映射策略。经证实,该算法始终能比顺序层次映射算法获得更高的整体视频接收质量,并具有高的带宽利用率和很小的复杂度。     

跳帧转码的运动矢量合成研究

由于终端设备、通信网络的异构性，视频服务需要提供跳帧转码功能，而直接使用输入码流运动矢量的视频跳帧算法导致视频质量下降严重．在前向主导运动矢量选择算法的基础上，讨论了运动矢量合成时运动矢量越界和主导帧内块不同处理方法对视频质量的影响，提出基于帧内刷新结构的前向主导运动矢量选择算法．实验结果表明，该算法可以有效阻断错误漂移，降低码率，提高视频质量．     

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.     

一种适用于H.264的时域差错掩盖改进算法

视频压缩码流在信道传输时 ,由于受到信道带宽或者稳定性的影响 ,容易发生数据的损坏或者丢失 ,这样不仅会对当前的视频帧产生影响 ,而且差错会延续到随后的视频帧 ,因此 ,需要采用某种技术来降低差错的影响。针对这一问题 ,在对最新视频压缩标准 H.2 6 4研究的基础上 ,基于 H.2 6 4标准的框架 ,对已有的差错掩盖算法进行了改进 ,提出了适合 H.2 6 4编码标准的时域子块匹配差错掩盖算法。该算法首先采用 8× 8的子块代替 16× 16的宏块 ,作为差错掩盖的运算单元 ,然后对不同的子块采用不同的边界像素 ,利用边界匹配算法 ,并通过改进的 1/ 4像素精度菱形搜索法在参考帧内找到最佳匹配块。实验结果证明 ,由于该算法有效地利用了 H.2 6 4压缩码流里的信息 ,因此 ,同传统的时域差错掩盖算法相比 ,对差错信号有更好的恢复效果。