一种基于信源信道联合的最优速率分配算法

针对视频数据在无线信道上可靠传输问题,提出了一种基于信源信道联合的最优速率分配算法。该算法在网络带宽一定的情况下,从信源、信道及差错弹性能力权衡考虑,引入了信源解码器的抗误码性能指标,根据不同的信道状态确定信源信道编码的最优速率分配方案,从而获得最大的可解码长度,并最终获取最佳重建视频质量。仿真结果表明,该方案与传统的联合信源信道速率分配算法相比可获得更高的性能增益,适合于视频数据在无线网络上传输。     

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。     

无线视频传输中MPEG-4自适应分包算法

MPEG-4视频流在无线信道中传输时，视频数据必须符合一定的结构，以便当信息丢失和比特错误时，能够进行差错控制。将视频净载荷按一定结构进行适当分组的处理称为打包或者分包。该文对分包的理论模型进行了分析，提出了一种兼顾封装效率和图像质量的自适应分包算法，并利用实时传输协议在GPRS无线信道中进行传输，取得了理想的视觉效果。     

An efficient bandwidth allocation algorithm for real-time VBR stream transmission under IEEE 802.16 wireless networks

This paper investigates variable rate control strategies for real-time multimedia variable bit rate (VBR) services over IEEE 802.16 broadband wireless networks. A data rate control mechanism is derived for the case where the uplink channel provides real-time services and the traffic rate parameter remains constant. This paper shows that the common queuing scheduling algorithms have some bandwidth allocation fairness problems for the real-time polling service (rtPS) in the MAC layer. In other words, the use of a VBR for the rtPS by a WiMAX system results in additional access latency jitter and bandwidth allocation disorder in the transmitted multimedia streams during the regular time interval polling of subscribe stations (SSs) for the contention bandwidth request period. However, the proposed scheduling algorithm solves these SSs contending with bandwidth resource allocation problems based on an extended rtPS (ertPS) of quality-of-service (QoS) pre-programming for a ranging response non-contention polling period. The adopted bandwidth allocation of max–min fairness queue scheduling uses a time constraint condition to transmit real-time multimedia VBR streaming in an IEEE 802.16 broadband wireless environment. In addition, we use the ns-2 simulation tool to compare the capacity of multimedia VBR stream and show that the proposed ertPS scheduling algorithm outperforms other rtPS scheduling algorithms.     

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。     

基于MPEG-4的综合抗误码方法

窄带高噪声的无线信道环境是视频通信的难点。在分析无线信道特点和移动设备有限的计算和存储能力的基础上,综合鲁棒性编码、解码端的错误隐藏等技术,设计了一种基于MPEG-4的综合抗误码方法（IERM）。该方法先将视频比特流划分为一些相对独立的“视频包”;然后将这些“视频包”按RTP协议封装成RIP包;最后在解码端实施有效的错误隐藏。经模拟实验表明：该方法能在窄带高误码的无线信道环境下,提高重建视频图像的视觉质量。     

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.     

一种基于视觉感知的H.264 码率控制算法

针对目前视频编码标准H.264 的码率控制算法未考虑人眼视觉感知、易导致编码 后视频图像质量波动的不足,提出了一种基于视觉感知的H.264 码率控制算法。首先,设计了 像素域的恰可察觉失真模型。在此基础上,根据各帧的恰可察觉失真的大小进行帧层比特分配。 其次,建立了基于结构相似度的率失真模型,并采用此模型设计了基本单元层(basic unit, BU) 的比特分配方案。最后结合二次速率-量化模型得到量化参数。实验结果表明,该算法与目前 H.264 中典型的码率控制算法相比,错误率降低了0.2%。     

一种改进IEEE802.15.3MAC协议机制的信道资源分配算法

本文简单介绍了IEEE802.15.3协议,分析了协议的MAC层QoS机制。针对实时VBR(VariableBitRate)业务流,提出了一种改进协议机制的信道资源分配算法,并在性能上与平均资源分配算法进行了分析和比较,通过仿真验证了此分配算法能有效降低实时VBR码流的丢帧率,提高系统传输的QoS性能。     

Cross-Layer prioritized H.264 video packetization and error protection over noisy channels

Video transmission over wireless channels is affected by channel-induced packet losses. Distortion due to channel errors can be alleviated by applying forward error correction. Aggregating H.264/AVC slices to form video packets with sizes adapted to their importance can also improve transmission reliability. Larger packets are more likely to be in error but smaller packets require more overhead. We present a cross-layer dynamic programming (DP) approach to minimize the expected received video distortion by jointly addressing the priority-adaptive packet formation at the application layer and rate compatible punctured convolutional (RCPC) code rate allocation at the physical layer for prioritized slices of each group of pictures (GOP). Some low priority slices are also discarded to improve protection to more important slices and meet the channel bit-rate limitations. We propose two schemes. Our first scheme carries out joint optimization for all slices of a GOP at a time. The second scheme extends our cross-layer DP-based approach to slices of each frame by predicting the expected channel bit budget per frame for live streaming. The prediction uses a generalized linear model developed over the cumulative mean squared error per frame, channel SNR, and normalized compressed frame bit budget. The parameters are determined over a video dataset that spans high, medium and low motion complexity. The predicted frame bit budget is used to derive the packet sizes and corresponding RCPC code rates for live transmission using our DP-based approach. Simulation results show that both proposed schemes significantly improve the received video quality over contemporary error protection schemes.     

基于改进LMS算法的TH-UWB接收机

跳时超宽带（TH-UWB）无线通信系统通常采用RAKE接收技术。而信道估计的准确度直接影响系统接收性能。提出一种基于梯度的变步长的LMS算法进行信道估计。与传统LMS算法相比,改进的LMS算法可以获得更小的MSE（Mean Square Error）从而为接收机提供更精确的信道估计量。同时,结果也表明该算法提高了整个接收机性能,并获得更小的BER（Bit Error Rate）。     

Visual sensitivity guided bit allocation for video coding

A video bit allocation technique adopting a visual distortion sensitivity model for better rate-visual distortion coding control is proposed in this paper. Instead of applying complicated semantic understanding, the proposed automatic distortion sensitivity analysis process analyzes both the motion and the texture structures in the video sequences in order to achieve better bit allocation for rate-constrained video coding. The proposed technique evaluates the perceptual distortion sensitivity on a macroblock basis, and allocates fewer bits to regions permitting large perceptual distortions for rate reduction. The proposed algorithm can be incorporated into existing video coding rate control schemes to achieve same visual quality at reduced bitrate. Experiments based on H.264 JM7.6 show that this technique achieves bit-rate saving of up to 40.61%. However, the conducted subjective viewing experiments show that there is no perceptual quality degradation. EDICS-1-CPRS, 3-QUAL.     

On Channel Capacity Estimation and Prediction for Rate-Adaptive Wireless Video

Packet drops caused by residue errors (MAC-layer errors) can severely deteriorate the wireless video quality. Prior studies have shown that this loss of quality can be circumvented by using forward error correction (FEC) to recover information from the corrupted packets. The performance of FEC encoded video streaming is critically dependent upon the choice of source and channel coding rates. In practice, the wireless channel conditions can vary significantly, thus altering the optimal rate choices. Thus, it is essential to develop an architecture which can estimate the channel capacity and utilize this estimate for rate allocation. In this paper we develop such a framework. Our contributions consist of two parts. In the first part we develop a prediction framework that leverages the received packets' signal to silence ratio (SSR) indications and MAC-layer checksum as side information to predict the operational channel capacity. In the second part, we use this prediction framework for rate allocation. The optimal rate allocation is dependent upon the channel capacity, the distribution of the (capacity) prediction error and the rate-distortion (RD) characteristics of the video source. Consequently, we propose a framework that utilizes the aforementioned statistics for RD optimal rate adaptation. We exhibit the efficacy of the proposed scheme by simulations using actual 802.11b wireless traces, an RD model for the video source and an ideal FEC model. Simulations using source RD models derived from five different popular video codecs (including H.264), show that the proposed framework provides up-to 5-dB improvements in peak signal-to-noise ratio (PSNR) when compared with conventional rate-adaptive schemes.      

Scene-Change Aware Dynamic Bandwidth Allocation for Real-Time VBR Video Transmission Over IEEE 802.15.3 Wireless Home Networks

IEEE 802.15.3, an emerging wireless technology, was designed to provide high-quality multimedia services at home. Dynamic bandwidth allocation for a multimedia connection should be considered in order to achieve higher channel utilization, less buffer and less delay, especially for variable bit rate (VBR) multimedia connections. For real-time VBR videos, the bandwidth requirement should be predicted adaptively for effective channel-time requests. Previously, the adaptive least-mean square (LMS) algorithm with fixed step size was applied to predict channel time requirements due to its simplicity and relatively good performance. However, the performance might degrade when scene changes occurred. In this paper, we modify the variable step-size LMS algorithm and apply it as our predictor (VSSNLMS) so that the prediction errors on scene changes can be effectively reduced. Using the prediction results of VSSNLMS, we propose a dynamic bandwidth-allocation scheme that is scene-change aware and can guarantee the delay bound of real-time VBR videos. Simulation results show that the VSSNLMS predictor is superior to previous LMS-type predictors in performance, and the proposed scheme has better performance in channel utilization, buffer usage, and packet loss     

H.264/SVC比特自适应分配的等级B帧码率控制算法

为减少码率控制中实际输出码率与目标码率之间的误差,改善视频序列编码尾部质量下降的缺陷,同时针对可伸缩视频编码中码率控制算法的不足,提出一种自适应比特分配的码率控制算法。算法基于对根据相邻图像帧之间的相关性以及对恒定质量的考虑,在图像组(GOP)之间平均分配目标比特,而在GOP内部则根据编码复杂度自适应分配目标比特,同时适当调整初始量化参数(QP),再根据目标比特分别计算P帧、B帧的QP。对不同的视频序列进行了实验测试,其结果验证了算法的准确性和有效性。     

Cycle-Based Rate Control for One-Way and Interactive Video Communications Over Wireless Channels

We propose a joint source-rate/channel-code control scheme for streaming video over a wireless channel. The scheme is designed to maximize the achievable source rate while guaranteeing an upper bound on the probability of starvation at the playback buffer. It can be applied to both one-way and interactive video communications. Rate control is performed adaptively on a per-cycle basis, where a cycle consists of a "good" channel period and the ensuing "bad" period. This cycle-based approach has two advantages. First, it reduces the fluctuations in the source bit rate, ensuring smooth variations in video quality. Second, it makes it possible to derive simple expressions for the starvation probability at the playback buffer, which we use to determine the optimal source rate and channel code for the good and bad periods of the subsequent cycle     

一种自适应改进无线TCP性能的算法

对采用TCP协议传输数据的实现过程及其在无线网络中可能遇到的问题进行了描述，针对这些问题提出了一种算法对无线链路下一时刻的误码率进行了估计，根据这个估计对TCP进行了修改以自适应选取合适的TCP报文段尺寸来传输数据。在网络仿真器（NS2）中仿真一个误码率时变的无线信道，把修改后的TCP置于NS仿真环境中进行仿真，实验结果表明了该改进方法有效。     

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.     

基于ATM的可变比特率视频传输

在目前采用的同步传输模式(STM)网络中,信道速率是固定的,这样由视频图像编码器产生的时变比特流无法直接接入固定比特率(CBR)信道。由于ATM网络所具有的异步传输,统计复用和支持QoS保证的特点,使之成为适合可变比特率(VBR)视频传输的网络技术。本文描述了VBR视频源的统计模型,论述了在ATM环境下的QoS控制、统计复用与带宽分配等问题。     

A minimum approximate-BER beamforming approach for PSK modulated wireless systems

A bearnforming algorithm is introduced based on the general objective function that approximates the bit error rate for the wireless systems with binary phase shift keying and quadrature phase shift keying modulation schemes. The proposed minimum approximate bit error rate （ABER） beamforming approach does not rely on the Gaussian assumption of the channel noise. Therefore, this approach is also applicable when the channel noise is non-Gaussian. The simulation results show that the proposed minimum ABER solution improves the standard minimum mean squares error beamforming solution, in terms of a smaller achievable system＇s bit error rate.