Joint rate control for VBR MPEG video on PVC ATM links

A joint rate control algorithm for variable bit rate (VBR) MPEG-compressed digital video on point-to-point permanent virtual circuit (PVC) ATM links is proposed. The algorithm controls the encoding mode of a number of video encoders that operate either on VBR or constant bit rate (CBR) mode. The algorithm selects the encoding mode based on the buffer occupancy of a multiplexer co-located with the encoders that interfaces them to the PVC link. VBR encoding is the predominant encoding mode used during congestion-free periods. CBR mode is used only during congested periods. When CBR is selected, new bit rates are jointly assigned to the encoders based on their relative encoding complexities. The bit rate assigned to the encoders are enforced by a CBR rate control local to each encoder. The performance of the joint rate control algorithm is evaluated through simulation of a packet multiplexer, where a number of connections are multiplexed onto a fixed-capacity channel. The performance is compared to that of multiplexing conventional CBR and open-loop VBR from the video quality and bandwidth efficiency points of view. Simulation results show that the proposed algorithm improves performance over multiplexed conventional CBR or open-loop VBR MPEG video, without significantly increasing implementation complexity. The application of the algorithm to video-on-demand over ATM is discussed.     

Low-delay rate control for real-time H.264/AVC video coding

This paper presents an efficient rate control scheme for the H.264/AVC video coding in low-delay environments. In our scheme, we propose an enhancement to the buffer-status based H.264/AVC bit allocation method. The enhancement is by using a PSNR-based frame complexity estimation to improve the existing mean absolute difference based (MAD-based) complexity measure. Bit allocation to each frame is not just computed by encoder buffer status but also adjusted by a combined frame complexity measure. To prevent the buffer from undesirable overflow or underflow under small buffer size constraint in low delay environment,the computed quantization parameter (QP) for the current MB is adjusted based on actual encoding results at that point. We also propose to compare the bits produced by each mode with the average target bits per MB to dynamically modify Lagrange multiplier (/spl lambda//sub MODE/) for mode decision. The objective of QP and /spl lambda//sub MODE/ adjustment is to produce bits as close to the frame target as possible, which is especially important for low delay applications. Simulation results show that the H.264 coder, using our proposed scheme, obtains significant improvement for the mismatch ratio of target bits and actual bits in all testing cases, achieves a visual quality improvement of about 0.6 dB on the average, performs better for buffer overflow and underflow,and achieves a similar or smaller PSNR deviation.     

Feedback priority control for MPEG video transmitted through ATM networks

Although ATM networks are suitable for transporting variable bit rate (VBR) traffic, bursty moving picture exports group (MPEG) traffic is tending to make the networks congested and results in cell loss. It is suggested that cells should be assigned different priorities according to the significance of the contained information such that the networks can try to protect the high-priority cells and obtain high video quality. Considering the payload behind the slice header, losing any cell would disable the decoding for the following part of the same slice. As a result, if any cell has to be discarded, the farther the position is, the lesser the following data are affected. In this article, a simple mechanism called ‘feedback control priority partition with RC-shaper (FBCPP&RC)’ is proposed to prevent losing cells at the former position by adaptively assigning high priority to these cells. The decision policy is simple and quick. It also employs an RC-shaper to smooth the burstiness of the cells so as to reduce cell-loss probability. The mechanism is expected to facilitate transmitting MPEG video as a cheaper available bit rate (ABR) traffic rather than VBR traffic. According to the simulation results, it performs well in improving the quality of single-frame as well as continuous pictures. It can also be applied to the new standard MPEG-II.     

Fuzzy-based learning rate determination for blind source separation

Many independent component analysis (ICA) algorithms have been proposed for blind source separation. These algorithms belong to the LMS-type algorithm in natural. Hence, the choice of the step-size reflects a tradeoff between misadjustment and the speed of convergence. Based on the separation state of outputs of the neural network for ICA, the paper develops a fuzzy inference-based step-size selection algorithm. The fuzzy inference system consists of two inputs (the second- and higher order correlation coefficients of output components) and one output (the fuzzy learning rate). In this way, the ICA algorithms become more efficient, which is verified by simulation results.     

Pyramid coding based rate control for constant bit rate video streaming

In this paper, a novel pyramid coding based rate control scheme is proposed for video streaming applications constrained by a constant channel bandwidth. To achieve the target bit rate with the best quality, the initial quantization parameter (QP) is determined by the average spatio-temporal complexity of the sequence, its resolution and the target bit rate. Simple linear estimation models are then used to predict the number of bits that would be necessary to encode a frame for a given complexity and QP. The experimental results demonstrate that the proposed rate control scheme significantly outperforms the existing rate control scheme in the Joint Model (JM) reference software in terms of Peak Signal to Noise Ratio (PSNR) and consistent perceptual visual quality while achieving the target bit rate. Finally, the proposed scheme is validated through experimental evaluation over a miniature test-bed.

     

无线自相似环境下的MPEG4码率自适应控制

对在无线局域网环境下,基于UDP的多媒体流MPEG4的码率自适应控制进行了研究,同时探讨了网络自相似等一些相关的问题,并借鉴通信原理中的自适应滤波方法,初步地解决了网络较平稳时码率收敛问题.通过在实验环境下对理论结果加以测试验证,实现了在自相似网络状况下,UDP协议的MPEG4码率自适应控制,该方案能更有效地跟踪实时网络有效带宽,比传统的模型方案更快速,更准确,大大地改善了服务质量.特别是采用带收敛特性的自适应算法后,当网络有效带宽趋于稳定时,媒体流输出速率的大小也收敛并趋于稳定,有利于Server和网络效率的改善.     

New bufferless rate control for high efficiency video coding

Multimedia Tools and Applications - As a crucial component of video compression, rate control regulates the encoding bitrates of compressed bitstreams to meet channel bandwidths while obtaining...     

Structural similarity-based rate control algorithm for 3D video

Multimedia Tools and Applications - Human vision easily identifies the structural degradation in a video and thus perceptual quality improvement is necessary. To adjust the video quality to match...     

机器人远程控制视频传输的实时性问题解决策略

分析了机器人远程控制中视频传输的延时问题。提出了一种混合的网络传输拓扑结构,并对传输速率的控制算法进行了移植和改进。利用NS模拟网络环境,在模拟网络中进行数据传输,验证了控制速率算法的可行性。试验结果表明在混合网络拓扑结构上使用控制速率的算法可以减少传输过程中的丢包率,提高视频传输的实时性。     

基于压缩域的关键帧快速提取方法

关键帧提取技术是基于内容检索和视频分析的基础。关键帧的使用减少了视频索引的数据量,同时也为视频摘要和检索提供了一个组织框架。首先介绍了目前的关键帧提取技术,然后提出了一种基于运动特征利用模糊推理算法从MPEG视频流中提取关键帧的方法。由于处理过程是直接从MPEG的压缩视频提取,不需对其解压,所以计算复杂度低,提高了提取速度。实验证明该方法效率高,可以比较好地代表视频内容。     

基于小波变换的MPEG视频流预测模型

提出了一种基于小波变换的MPEG视频流的预测模型,该模型对原始数据做Haar小波变换以后,充分考虑尺度系数和乘子的不同统计特性,对尺度系数和乘子采用不同的预测模型分别预测,最终通过多重分形小波模型建模过程合成MPEG视频流序列。利用仿真和数学分析的方法对预测模型在刻画MPEG视频流的主要统计特性方面进行性能分析,从误差、数据的概率分布特性以及自相似特性等方面,证实了该预测模型的优越性能。     

FJND-based fuzzy rate control of scalable video for streaming applications

Multimedia Tools and Applications - A fuzzy rate controller with buffer constraint in combination with a perceptual quality controller is proposed for streaming applications of the AVC/H.264...     

Region-based rate control and bit allocation for wireless video transmission

In this paper, we propose a joint source-channel region-based rate control algorithm for real-time video transmissions over wireless systems. During the video transmission, the channel throughput available to the video encoder in the wireless systems is inherently variable, due to the retransmission of the error packets using the automatic repeat request (ARQ) error control. The variable data rate of the wireless system is characterized by the packet-level Gilbert two-state Markov Model, the parameters of which are extracted from the statistical properties of the channel information obtained from the wireless channel simulator. The proposed algorithm adopts a fast but effective block-based segmentation method to extract the regions of interest. Unlike traditional bit allocation methods used in the region/content-based rate control, the algorithm exploits the most effective criteria "coding qualities" as quantitative factors to directly control bit allocation among different regions so as to achieve better visual quality in the regions of interest. The computational complexity of the algorithm is low making it suitable for real-time applications. Compared with the MPEG-4 rate control algorithm, our algorithm can effectively enhance the perceptual quality for the regions of interest and significantly reduce the number of frame skipping; thereby, improve the smoothness of the video.     

Feature-based real-time video stabilization for vehicle video recorder system

This paper presents a fast and effective method based on features to obtain real-time video stabilization for vehicle video recorder system. The corresponding feature points are first obtained from two consecutive frames and then optical flows are calculated based on these points. Next, the obtained optical flows are mapped to polar coordinates to obtain clusters and remove incorrect optical flows. These obtained clusters are used to evaluate the global motion and rotation angle. Finally, the obtained global motion and rotation angle are smoothed and then compensated to obtain the stabilized video. Experimental results show that the proposed method has good performance for video stabilization.     

Real-time constrained TCP-compatible rate control for video over the Internet

This paper describes a rate control algorithm that captures not only the behavior of TCP's congestion control avoidance mechanism but also the delay constraints of real-time streams. Building upon the TFRC protocol , a new protocol has been designed for estimating the bandwidth prediction model parameters. Making use of RTP and RTCP, this protocol allows to better take into account the multimedia flows characteristics (variable packet size, delay ...). Given the current channel state estimated by the above protocol, encoder and decoder buffers states as well as delay constraints of the real-time video source are translated into encoder rate constraints. This global rate control model, coupled with an H.263+ loss resilient video compression algorithm, has been extensively experimented with on various Internet links. The experiments clearly demonstrate the benefits of 1/ the new protocol used for estimating the bandwidth prediction model parameters, adapted to multimedia flows characteristics, and of 2/ the global rate control model encompassing source buffers and end-to-end delay characteristics. The overall system leads to reduce significantly the source timeouts, hence to minimize the expected distortion, for a comparable usage of the TCP-compatible predicted bandwidth.     

Efficient MPEG compressed video analysis using macroblock typeinformation

Efficient indexing methods are required to handle the rapidly increasing amount of visual information within video databases. Video analysis that partitions the video into clips or extracts interesting frames is an important preprocessing step for video indexing. We develop a novel method for video analysis using the macroblock (MB) type information of MPEG compressed video bitstreams. This method exploits the comparison operations performed in the motion estimation procedure, which results in specific characteristics of the MB type information when scene changes occur or some special effects are applied. Only a simple analysis on MB types of frames is needed to achieve very fast scene change, gradual transition, flashlight, and caption detection. The advantages of this novel approach are its direct extraction from the MPEG bitstreams after VLC decoding, very low complexity analysis, frame-based detection accuracy and high sensitivity     

Modeling video sources for real-time scheduling

What is the impact of the autocorrelation of variable-bit-rate (VBR) sources on real-time scheduling algorithms? Our results show that the impact of long term, or interframe, autocorrelation is negligible, while the impact of short term, or intraframe, autocorrelation can be significant. Such results are essentially independent of the video coding scheme employed. To derive these results, video sequences are modeled as a collection of stationary subsequences called scenes. Within a scene, a statistical model is derived for both the sequence of frames and of slices. The model captures the distribution and the autocorrelation function of real-time video data. In previous work, the pseudoperiodicity of the slice level auto-correlation function made it difficult to develop a simple yet accurate model. We present a generalization of previous methods that can easily capture this pseudoperiodicity and is suited for modeling a greater variety of autocorrelation functions. By simply tuning a few parameters, the model reproduces the statistic behavior of sources with different types and levels of correlation on both the frame and the slice level.     

Impact of video scheduling on bandwidth allocation for multiplexed MPEG streams

We present efficient schemes for scheduling the delivery of variable-bit-rate MPEG-compressed video with stringent quality-of-service (QoS) requirements. Video scheduling is being used to improve bandwidth allocation at a video server that uses statistical multiplexing to aggregate video streams prior to transporting them over a network. A video stream is modeled using a traffic envelope that provides a deterministic time-varying bound on the bit rate. Because of the periodicity in which frame types in an MPEG stream are typically generated, a simple traffic envelope can be constructed using only five parameters. Using the traffic-envelope model, we show that video sources can be statistically multiplexed with an effective bandwidth that is often less than the source peak rate. Bandwidth gain is achieved without sacrificing the stringency of the requested QoS. The effective bandwidth depends on the arrangement of the multiplexed streams, which is a measure of the lag between the GOP periods of various streams. For homogeneous streams, we give an optimal scheduling scheme for video sources at a video-on-demand server that results in the minimum effective bandwidth. For heterogeneous sources, a sub-optimal scheduling scheme is given, which achieves acceptable bandwidth gain. Numerical examples based on traces of MPEG-coded movies are used to demonstrate the effectiveness of our schemes.