A new rate control scheme for H.263 video transmission

In some image/video applications, the variable bit rate image/video bitstream will be transmitted over a constant bit rate transmission channel, in which a channel buffer is employed. In this study, a new rate control scheme for H.263 video transmission is proposed. Three proposed techniques include: preprocessing the INTRA coded macroblock (MB) incorporated the just-noticeable-distortion (JND) concept, constructing a bit estimation model in the frequency domain (instead of the bit estimation model in the spatial domain employed in Test Model Near-term version 11 (TMN11)), and adjusting quantization parameter (QP) for each MB by a Lagrangian optimization strategy.In the proposed approach, the target number of bits for a video frame is first obtained by using a simple rate control procedure for the frame layer. The proposed JND preprocessing is applied on all the INTRA coded MBs so that the number of coded bits for a scene changed frame will decrease, without any perceptual loss. Within the MB layer, instead of the bit estimation model in the spatial domain employed in TMN11, a bit estimation model in the frequency domain, directly depending on discrete cosine transform coefficients, is proposed. Then the Lagrange multiplier is used to determine the optimal QP for each MB. The resulting QP and number of coded bits of the current MB are fed backward to update the parameters of the bit estimation model.Based on the simulation results obtained in this study, the proposed approach can meet the target bit rate more accurately, keep a lower channel buffer fullness (delay), and have a larger average frame rate than TMN11, whereas the peak signal-to-noise ratio value and the processing time of the proposed approach are “approximately” as good as that of TMN11.     

基于统计建模的VVC快速码率估计算法

为降低新一代通用视频编码（versatilevideocoding,VVC）标准率失真优化过程的编码复杂度,提出一种基于统计建模的快速码率估计算法。首先,算法充分考虑依赖性量化（dependent quantization,DQ）的量化行为和熵编码中的上下文依赖,提出可以准确刻画编码过程中上下文状态迁移的码率特征,初步预估变换单元（transformunit,TU）中部分语法元素的码率;其次,基于系数分布特性,定义系数混乱度特征和稀疏度特征来区分系数分布差异带来的码率影响,并构建TU级码率模型;最后,算法根据码率构成特性将大尺寸TU和小尺寸TU分开建模实现更精准的码率预估。通过统计方式对大量样本进行回归训练,得到最终的线性码率模型,并应用于VVC的模式决策中。实验结果表明,所提出算法在随机访问（random access,RA）配置下,可以实现16.289%的复杂度降低,而码率变化率（Bjontegaard delta bit rate,BD-BR）仅增加1.567%。     

一种低复杂度的HEVC码率控制算法

针对HEVC视频编码中帧层比特分配和率失真模型参数自适应更新,提出了一种改进的码率控制算法.该算法引入牛顿法来自适应更新率失真模型的参数;同时提出一种帧层复杂度度量方法,利用图像自身的内容特性来调整帧层比特分配;最后在编码过程中选择合适的量化参数.实验结果表明:与参考算法比较,该算法能大大降低计算复杂度,同时保持良好的码率控制性能,对含有场景突变的视频改进效果尤为明显.     

Novel distribution model of transformed coefficients in video coding using quad-tree structured block partitioning

Today’s video coding standard such as high efficiency video coding uses a full quad-tree structured block partitioning, so the underlying statistics of transformed coefficients becomes more complicated to estimate than the previous standards due to the coding structure. However, a statistical distribution of transformed residue is important for a design of a smart encoder. Thus, in this paper, we present a theoretic analysis of a distribution of transformed coefficients produced from an encoder using different transform sizes, and derive a probability density function (pdf) for the estimation. The proposed density model provides a more accurate distribution model than the conventional pdfs. Parameters are theoretically estimated, and rate-distortion model is established from the proposed pdf. We also apply the proposed method to a rate control problem to show the efficiency of the proposed density model. Our experimental results show that the proposed method is better capable of modeling the mixed sources of multiple-type transform coefficients occurred from the quad-tree coding structure of transform and provides an accurate estimate in rate control.

     

视频复杂度自适应的H.264基本单元层码率控制方法

为了提高JVT-H017码率控制提案中的基本单元层比特数分配方法在视频序列复杂度较高时分配的准确度,提出了一种视频复杂度自适应的基本单元层码率控制方法。利用DMAD参数来判断视频场景的复杂度,对JVT-H017的比特数分配方法进行了改进;同时提出了一种基本单元层编码的理想情况,并以此理想情况下的比特数消耗情况来对基本单元的比特数进行修正。仿真结果表明,该方法有效地提高了视频编码的峰值信噪比(PSNR),最大可提高0.49 dB,并且输出码率更接近预定编码码率。     

Blind DOA and polarization estimation for polarization-sensitive array using dimension reduction MUSIC

A novel blind direction-of-arrival (DOA) and polarization estimation algorithm for polarization-sensitive uniform linear array using dimension reduction multiple signal classification (MUSIC) is proposed in this paper. The proposed algorithm utilizes the signal subspace to obtain an initial estimation of DOA, then estimates more accurate DOA through a one-dimensional (1-D) local searching according to the initial estimation of DOA, and finally obtains polarization parameter estimation via the estimated polarization steering vectors. The proposed algorithm, which only requires a one-dimension local searching, can avoid the high computational cost within multi-dimensional MUSIC algorithm. The simulation results reveal that the proposed algorithm has better DOA and polarization estimation performance than both estimation of signal parameters via rotational invariance technique algorithm and trilinear decomposition algorithm. Furthermore, the proposed algorithm can be suitable for irregular array geometry, obtain automatically paired multi-dimensional parameter estimation, and avoid multi-dimensional searching. Simulation results verify the effectiveness of the proposed algorithm.     

一种基于ρ域的H．264／AV码率控制算法

提出了一种基于ρ域线性率失真模型的H．264／AVC码率控制方法。算法主要利用ρ域线性率失真模型来解决量化参数同时用于码率控制和率失真优化的矛盾。同时,用非零运动向量个数来估计宏块的头比特数,进而解决头比特的合理估计问题。实验通过对新算法和JVT—H017算法的比较,得到本文算法不但可以获得更精确的码率控制,而且使输出码率更加的平稳,本文算法实现相对简单,适用于实时视频通信。     

信号DOA和极化参数的自适应估计

运用ESPRIT算法进行信号参数的估计,关键是对信号子空间的估计。在实际运用中,接收到的信号的参数往往是随时间变化的,因此要得到信号参数的实时估计值就需要根据阵列的接收数据对信号子空间进行更新。分析了如何运用ESPRIT算法得到信号的DOA和极化参数,并在矩阵扰动理论的基础上,利用矩阵特征分解一阶修正方法更新特征值和特征向量,从而使得ESPRIT算法能够自适应地估计信号DOA和极化参数。最后通过Matlab仿真验证了该方法的有效性。     

Frame-level bit allocation based on incremental PID algorithm and frame complexity estimation

Since the current rate control schemes in H.264 do not have the capability of efficient frame-level bit allocation, the video quality varies significantly from frame to frame especially for sequences with sudden scene changes or high motion activities. To overcome the limitation of frame-level bit allocation, we improve H.264 rate control scheme using two tools, the incremental proportional–integral–differential (PID) algorithm and the frame complexity estimation. The incremental PID algorithm is first introduced to control the buffer and reduce the influence of the buffer abrupt fluctuation in the process of frame-level bit allocation. To reduce more video quality variations, the frame target bit allocation is also adjusted by frame complexity that is estimated by residual energy. Simulation results show that the proposed rate control scheme, without introducing expensive computational complexity, decreases the average standard deviation of video quality by 32.29%.     

基于独立混合模型的EM算法参数初始化实现方法

隐马尔科夫树( Hidden Markov Tree, HMT )的状态不能被观测到,只能观测到另一个与状态有联系的量,通过观测量估计HMT模型参数是一个不完全数据参数估计问题。期望最大化( Expectation Maximization, EM )算法是一种求参数极大似然估计的迭代算法,可以用于解决不完全数据参数估计问题,因此被广泛应用于HMT模型的参数估计中。当初始参数偏离真实参数较大时,EM算法迭代次数多,收敛速度慢,通过一个计算量不大的参数初始化处理,能够有效减少EM算法的迭代次数,加快收敛速度。本文提出了一种基于独立混合模型的参数初始化方法,详细介绍了该方法的实现过程,通过采用独立混合模型进行参数初始化,使得EM算法的迭代次数明显减少,收敛速度大大提高。最后,计算机仿真验证了该方法的可行性和有效性。      

New rate-distortion modeling and efficient rate control for H.264/AVC video coding

Rate control (RC) is crucial in controlling compression bit rates and qualities for networked video applications. In this paper, we propose a new rate-distortion (R-D) model and an efficient rate control scheme for H.264/AVC video coding, which elegantly resolve the inter-dependency problem between rate-distortion optimization and rate control by eliminating the need of coding complexity prediction for an inter-frame. The objective is to achieve accurate bit rate, obtain optimal video quality while reducing quality variations and simultaneously handling buffer fullness effectively. The proposed algorithm encapsulates a number of new features, including a coding complexity measure for intra-frames, a rate-distortion model, an accurate quantization parameter (QP) estimation for intra-frames, an incremental quantization parameter calculation method for inter-frames, a proportional+integral+derivative (PID) buffer controller, and an intelligent bit-allocation-balancing technique. Our experimental results demonstrate that the proposed scheme outperforms the JVT-G012 solution by providing accurate rate regulation, effectively reducing frame skipping, and finally improving coding quality by up to 1.80 dB.     

Content-adaptive bitstream-layer model for coding distortion assessment of H.264/AVC networked video

Bitstream-layer models are designed to use the information extracted from both packet headers and payload for real-time and non-intrusive quality monitoring of networked video. This paper proposes a content-adaptive bitstream-layer (CABL) model for coding distortion assessment of H.264/AVC networked video. Firstly, the fundamental relationship between perceived coding distortion and quantization parameter (QP) is established. Then, considering the fact that the perceived coding distortion of a networked video significantly relies on both the spatial and temporal characteristics of video content, spatial and temporal complexities are incorporated in the proposed model. Assuming that the residuals before Discrete Cosine Transform (DCT) keep to the Laplace distribution, the scale parameters of the Laplace distribution are estimated utilizing QP and quantized coefficients on the basis of the Parseval theorem firstly. Then the spatial complexity is evaluated using QP and the scale parameters. Meanwhile, the temporal complexity is obtained using the weighted motion vectors (MV) considering the variations in temporal masking extent for high motion regions and low motion regions, respectively. Both the two characteristics of video content are extracted from the compressed bitstream without resorting to a complete decoding. Using content related information, the proposed model is able to adapt to different video contents. Experimental results show that the overall performance of CABL model significantly outperforms that of the P.1202.1 model and other coding distortion assessment models in terms of widely used performance criteria, including the Pearson Correlation Coefficient (PCC), the Spearman Rank Order Correlation Coefficient (SROCC), the Root-Mean-Squared Error (RMSE) and the Outlier Ratio (OR).     

一种新的视频编码二次率失真模型及其性能分析

基于对视频编码比特率与内容复杂度之间关系的研究,该文提出了一种新的视频编码二次率失真模型。该模型保留泰勒级数中的常数项,并通过合理引入内容复杂度,提高了描述实际信源率失真特性的准确性。大量实验分析表明该模型具有良好的性能,可广泛应用于各种速率控制算法以及其他使用率失真模型的场合,以提高视频编码器的率失真性能。     

分布式视频编码中拉普拉斯-柯西混合分布相关噪声模型研究

该文通过研究变换域分布式视频编码中原始Wyner-Ziv(WZ)帧与相应边信息的残差系数特性,发现大残差和小残差系数统计分布与传统的拉普拉斯分布存在一定偏差。为了减少这种差异,提出一种拉普拉斯-柯西混合分布(LCMD)相关噪声模型及其参数估计算法。该混合模型利用改进的拉普拉斯分布描述小残差系数的分布,采用柯西分布描述大残差系数。实验结果表明该文提出的混合模型能较精确地描述WZ帧和边信息间的残差系数分布,从而有效地改善了变换域分布式视频编码的率失真性能,并减少系统解码端计算复杂度。     

高清视频联网系统中一种新的丢包率分析方法

为了提高系统性能并最大限度降低算法实现复杂度,本文提出了一种新的高清视频联网系统中丢包率的更优近似分析方法,重点研究包长度对丢包率的影响。首先使用马尔可夫链模型估算数据包的误码率,进而得出数据包网络时延的概率分布,最后计算出基于包长度的丢包率的理论表达式。仿真结果表明,通过减小发包大小可以有效降低丢包率,同时基于包长度的丢包率的理论分析方法可以有效地运用到高清视频联网系统的分析研究中。     

Fast detection and parameter estimation for polyphase-code radar signal

The detection and parameter estimation for polyphase-code radar signal are analysed in this article. In view of the fact that traditional algorithms of signal detection and parameter estimation have enormous computational complexity, a joint fast algorithm employed fractional operation is proposed to detect the polyphase-code radar signal and estimate its modulation parameter. The proposed algorithm firstly detects the signal and estimates the sweep rate with the detection statistics derived from fractional autocorrelation. Then proposed algorithm achieves the estimation of other modulation parameters by using fractional Fourier transform (FrFT). Simulation results have verified the effectiveness of the proposed algorithm. Compared with traditional algorithms, proposed algorithm has identical performance of detection and parameter estimation, but can remarkably reduce computational cost. The proposed algorithm is suitable for the application of practical equipment.     

SVD-based modeling for image texture classification using wavelet transformation.

This paper introduces a new model for image texture classification based on wavelet transformation and singular value decomposition. The probability density function of the singular values of wavelet transformation coefficients of image textures is modeled as an exponential function. The model parameter of the exponential function is estimated using maximum likelihood estimation technique. Truncation of lower singular values is employed to classify textures in the presence of noise. Kullback-Leibler distance (KLD) between estimated model parameters of image textures is used as a similarity metric to perform the classification using minimum distance classifier. The exponential function permits us to have closed-form expressions for the estimate of the model parameter and computation of the KLD. These closed-form expressions reduce the computational complexity of the proposed approach. Experimental results are presented to demonstrate the effectiveness of this approach on the entire 111 textures from Brodatz database. The experimental results demonstrate that the proposed approach improves recognition rates using a lower number of parameters on large databases. The proposed approach achieves higher recognition rates compared to the traditional sub-band energy-based approach, the hybrid IMM/SVM approach, and the GGD-based approach.     

The transmission of MPEG‐2 VBR video under usage parameter control

This paper studies the transmission of MPEG‐2 VBR video over ATM network under usage parameter control. The idea is to seek a compromise between the network utilization and the quality of video service by applying UPC‐based rate control strategies to the video source. A modified leaky bucket algorithm is proposed to calculate the constraints on the bit‐rate guaranteeing conformance to peak cell rate, sustainable cell rate and burst tolerance usage parameters. Two rate control strategies, one for real‐time generated video coding and the other for pre‐compressed video, are proposed for MPEG‐2 VBR video. The rate control strategies control the video source to generate traffic conforming to the constraints on the bit rate. The experimental results show that both the UPC‐based rate control strategies can provide lossless transmission from the source perspective as well as to reduce the burstiness of the traffic. To keep within the bit‐rate allowed, the control method uses coarser quantization to maintain better picture quality than that by removing the number of AC transformed coefficients. The slight degradation of picture quality caused by the source rate control is preferable than the severe drop of picture quality caused by the cell loss at UPC. Copyright © 2001 John Wiley & Sons, Ltd.     

Rate control of MPEG video for consistent picture quality

We propose a rate control scheme using a rate-distortion (R-D) estimation model, which produces a consistent picture quality between consecutive frames. Our R-D estimation method offers a closed-form mathematical model that enables us to predict the bits and the distortion generated from a frame encoded at a given quantization parameter (QP) and vice versa. Its most attractive feature is its low computational complexity. Furthermore, it is accurate enough to be applied to practical video coding. In our simulation, the estimation errors for rate and distortion are less than 2.5% and 1.5%, respectively. Therefore, the proposed rate control scheme is appropriate for applications requiring low delay, low complexity, and the ability to control output bit-rate and quality accurately. Our scheme ensures that the video buffers do not underflow or overflow by satisfying the buffer constraint, and it also prevents quality difference between consecutive frames from exceeding a certain demanded level by adopting a distortion constraint. In addition, a consistent picture quality is maintained within a frame, and error propagation, caused by quality degradation of anchor frames, is reduced by differentiating the control procedure for anchor frames from that for nonanchor frames. Simulation results show that our control scheme achieves 0.52-1.84 dB peak signal-to-noise ratio (PSNR) gain over MPEG-2 Test Model 5 (TM5) rate control and maintains very consistent quality within a frame as well as between frames.