一种甚低速率语音编码的抗误码算法

为满足在高误码率的窄带信道上进行语音通信的需求，本文研究了一种适用于甚低速率语音通信的抗误码参数估值算法。在一定的解码状态下，声码器通过计算最小均方误差（MMSE）估计的方法估计最优参数，充分降低信道误码对重建语音质量的影响。对于解码状态参数，通过计算最大后验转移概率的方法作最佳估计，并给出了一种简化的计算方法。这种抗误码算法复杂度低。计算机仿真结果表明，在不同误码率下该算法恢复出的语音的平均谱失真（ASD）低于帧删除方法的谱失真最大可达0.4dB。     

Goodput analysis and link adaptation for IEEE 802.11a wireless LANs

Link adaptation to dynamically select the data transmission rate at a given time has been recognized as an effective way to improve the goodput performance of the IEEE 802.11 wireless local-area networks (WLANs). Recently, with the introduction of the new high-speed 802.11a physical layer (PHY), it is even more important to have a well-designed link adaptation scheme work with the 802.11a PHY such that its multiple transmission rates can be exploited. In this paper, we first present a generic method to analyze the goodput performance of an 802.11a system under the distributed coordination function (DCF) and express the expected effective goodput as a closed-form function of the data payload length, the frame retry count, the wireless channel condition, and the selected data transmission rate. Then, based on the theoretical analysis, we propose a novel MPDU (MAC protocol data unit)-based link adaptation scheme for the 802.11a systems. It is a simple table-driven approach and the basic idea is to preestablish a best PHY mode table by applying the dynamic programming technique. The best PHY mode table is indexed by the system status triplet that consists of the data payload length, the wireless channel condition, and the frame retry count. At runtime, a wireless station determines the most appropriate PHY mode for the next transmission attempt by a simple table lookup, using the most up-to-date system status as the index. Our in-depth simulation shows that the proposed MPDU-based link adaptation scheme outperforms the single-mode schemes and the autorate fallback (ARF) scheme-which is used in Lucent Technologies' WaveLAN-II networking devices-significantly in terms of the average goodput, the frame drop rate, and the average number of transmission attempts per data frame delivery.     

A Fast Intra Skip Detection Algorithm for H.264/AVC Video Encoding

A fast intra skip detection algorithm based on the rate‐distortion (RD) cost for an inter frame (P‐slices) is proposed for H.264/AVC video encoding. In the H.264/AVC coding standard, a robust rate‐distortion optimization technique is used to select the best coding mode and reference frame for each macroblock (MB). There are three types of intra predictions according to profiles. These are 16×16 and 4×4 intra predictions for luminance and an 8×8 intra prediction for chroma. For the high profile, an 8×8 intra prediction has been added for luminance. The 4×4 prediction mode has 9 prediction directions with 4 directions for 16×16 and 8×8 luma, and 8×8 chrominance. In addition to the inter mode search procedure, an intra mode search causes a significant increase in the complexity and computational load for an inter frame. To reduce the computational load of the intra mode search at the inter frame, the RD costs of the neighborhood MBs for the current MB are used and we propose an adaptive thresholding scheme for the intra skip extraction. We verified the performance of the proposed scheme through comparative analysis of experimental results using joint model reference software. The overall encoding time was reduced up to 32% for the IPPP sequence type and 35% for the IBBPBBP sequence type.     

On the complexity reduction of turbo decoding for wideband CDMA

Two simple but effective methods for reducing the average complexity (and power consumption) of the conventional turbo-cyclic-redundancy-check decoding scheme with negligible performance degradation in a wideband direct-sequence code-division multiple-access (W-CDMA) environment are introduced. When applied to a W-CDMA turbo code with frame length 640 b at a bit-error rate (BER) of 10/sup -6/, the resultant modified schemes can save up to 73% of the average decoding complexity, relative to the conventional scheme. In general, the proposed schemes are more attractive for short-frame and low-BER applications.     

一种基于奇偶校验码级联极化码的低复杂度译码算法

极化码作为一种纠错码,具有较好的编译码性能,已成为 5G 短码控制信道的标准编码方案。但在码长较短时,其性能不够优异。作为一种新型级联极化码,奇偶校验码与极化码的级联方案提高了有限码长的性能,但是其译码算法有着较高的复杂度。该文针对这一问题,提出一种基于奇偶校验码级联极化码的串行抵消局部列表译码(PC-PSCL)算法,该算法在编码前进行外码构造,通过高斯近似(GA)得到的子信道错误概率选取较不可靠的信息位,对选取的较不可靠的信息位进行串行抵消列表(SCL)译码和奇偶校验,其余信息比特仅进行串行抵消(SC)译码。仿真结果表明,在高斯信道下,当码长为512,码率为1/2,误帧率为10–3,最大列表长度为8时,该文提出的低复杂度译码算法比SCL译码算法获得了0.5 dB的增益;与基于奇偶校验的SCL译码算法性能相近,但是空间复杂度和时间复杂度分别降低了38.09%, 15.63%。     

基于退避预测的ACB动态接入新方案

在LTE-A网络的过载场景中,机器类通信（Machine Type Communication,MTC）设备的突发性接入会使得网络发生严重的拥塞,甚至死锁,造成网络的接入效率低下.在可用前导资源有限的前提下,根据实时负载数控制发起接入的设备数可以有效降低前导的碰撞概率,但是控制方法尚不明确.为此,本文提出了一种接入类别限制（Access Class Barring,ACB）的动态接入机制来优化海量MTC的随机接入性能.建立了一种基于退避预测的估计模型,该模型根据重传的设备数和状态转移过程估计出了实时活跃的设备数.结合估计模型和ACB参数调整可以最优化实时成功接入的设备数,能够有效地提高设备的接入成功率.本文在不同负载强度场景下,将提出的ACB动态接入机制和现有的动态ACB机制的接入性能进行了比较.仿真结果证明,本文提出的ACB动态接入机制的接入成功率为100%.而且,与现有的ACB动态接入机制相比,所提的新方案的平均接入时延更低.     

Improving goodput in IEEE 802.11 wireless LANs by using variable size and variable rate (VSVR) schemes

In IEEE 802.11 wireless LANs, channel quality, network load, as well as the protocol itself are time‐varying, limiting the goodput performance improvement in wireless LANs. Therefore, it becomes critical to dynamically adjust parameters of MAC and PHY layers according to variations of channel quality. In this paper, we propose variable frame size and variable data rate schemes for goodput enhancement. We first propose two optimal frame size predictors: a goodput regulator to maintain the committed goodput for non‐greedy applications and an optimal frame size predictor for maximizing the goodput for greedy applications. Then, we propose a data rate drafting scheme and develop a variable size and variable rate (VSVR) scheme for further goodput improvement. Our extensive simulation results show that the proposed VSVR algorithm can double the channel goodput of current implementations. Moreover, the proposed scheme can be easily integrated with the current implementations of the wireless LAN MAC protocol. Copyright © 2004 John Wiley & Sons, Ltd.     

TOA/RSS混合信息室内可见光定位方法

为提高室内定位精度,提出一种基于混合到达时间/接收信号强度(TOA/RSS)信息的定位方法。针对室内可见光定位中存在的多径效应造成的定位非线性误差,引入前置无迹卡尔曼滤波的粒子滤波算法,将TOA信息与RSS信息相融合,达到修正非线性误差的目的。然后综合考虑接收端惯性传感参数,对接收端进行运动分析,提升估算坐标的精度。在长宽均为5 m、高度为3 m的室内进行定位仿真,在12 W发光二极管(LED)发射功率下,所提方法获得了平均定位误差为2.02 cm的定位精度。仿真结果证明,所提定位方法的定位性能总体优于指纹定位方法和三边定位的RSS定位方法,具有较强的鲁棒性和较低的定位延迟。     

动态TDD技术HARQ时序冲突的解决方案

动态TDD技术的引入不仅能够很好地解决突发性业务需求增长较快的问题,还能提升小区通信系统的服务质量.在动态TDD系统中,无线帧配置可以通过系统中的负载量来改变.但是一些子帧可能会在重配后改变传输方向,导致一些HARQ进程出现时序冲突,影响系统性能.为了解决动态TDD系统中的HARQ时序冲突问题,提出了一种基于过渡无线帧的混合自动重传方案.当重配周期到来时,在重配点后增加一个过渡无线帧,使得HARQ时序以一种较简单、时序冲突较少的方式过渡到重配后的无线帧.在LTE系统仿真平台上建立HARQ模型进行性能验证,仿真结果表明,所提方案能有效地提升系统资源利用率,并给系统平均吞吐量带来增益.     

基于多LED的高精度室内可见光定位方法

针对可见光室内定位问题,该文基于接收信号强度(RSS)定位技术,提出一种利用多个LED发射端实现室内定位的方法,即MLED-RSS定位算法。该方法在充分考虑LED拓扑结构对定位性能影响的基础上,利用部署在室内的多个LED,合理选择其中3个LED作为发射节点,采用改进的三边定位法获得定位目标位置信息。定位算法可以有效地解决可见光定位存在的遮挡效应。仿真实验表明,MLED-RSS算法可以实现高定位精度。     

Dual-mode SM/STBC system with antenna subset selection employing ML detection

Multiple input multiple output (MIMO) antenna system is a promising candidate to meet the demands of 4th Generation (4G) cellular communication systems by offering increased spectral efficiency through the spatial multiplexing (SM) gain, and improved link reliability through the space–time block coding (STBC) diversity gain. This paper presents a new scheme that combines the dual-mode SM/STBC and the antenna subset selection (AnSS) schemes. In the proposed scheme, the combination of the SM/STBC switching and the full antenna subset selection (AnSS) at both the transmitter (Tx) and the receiver (Rx) ends of the communication channel are adaptively selected through a simple algorithm based on the singular values of the channel matrix at the Rx side. Thus, the new scheme achieves the best BER performance over the previous works regardless of the data rate. The simulation results show that the proposed scheme with the full AnSS outperforms the previous works, by up to the 12.5 dB at the bit error rate (BER) of 10^‐5${10}^{‐5}$. Further, a partial AnSS is also proposed which dramatically reduces both the computational complexity (by 31%) and the hardware (by 50%), cost, without any appreciable loss in the BER performance, when compared with the full AnSS.     

一种H.264视频流自适应率失真优化编码算法

为了提高编码视频流在丢包网络环境中的抗误码性能,目前比较常用的是采用帧内刷新算法.在率失真框架之内的帧内优化编码刷新算法,则被认为是更为直接和有效的解决办法.在视频编码标准H.264/JVT中采用的就是这种算法.然而由于没有考虑到信道丢包率对编码器端进行仿真解码次数的影响,从而导致在进行率失真优化编码时的计算量较大,编码耗时较长,严重影响了编码器的编码效率.基于以上分析,提出了一种改进的自适应率失真优化编码算法.将H.264标准率失真优化编码算法中计算解码器端视频帧期望失真度的代数平均值算法,改进为加权平均值算法.仿真实验表明,提出的算法可根据信道丢包率和模拟信道状态个数信息来自适应地决定编码器端进行仿真解码计算的次数,从而有效降低H.264标准率失真优化编码算法中的计算冗余和计算复杂度,节省编码耗时.在模拟信道状态个数默认为30个时,本算法最多可节省近55%的编码耗时.     

Frame-level traffic splitting for link aggregation in data center networks

Ethernet link aggregation, which provides an easy and cost-effective way to increase both bandwidth and link availability between a pair of devices, is well suited for data center networks. However, all the traffic splitting algorithms used in existing Ethernet link aggregation are flow-level which do not work well owing to the traffic characteristics of data centers. Though frame-level traffic splitting can achieve optimal load balance and the maximum benefits from aggregated capacity, it is generally deprecated in most cases because of frame disordering which can disrupt the operation of many Internet protocols, most notably transmission control protocol (TCP). To address this issue, we first investigate the causes of frame disordering in link aggregation and find that all of them either are no longer true or can be prevented in data centers. Then we present a byte-counter frame-level traffic splitting algorithm which achieves optimal performance while causes no frame disordering. The only requirement is that frames in a flow are the same size which can be easily met in data centers. Simulation results show that the proposed frame-level traffic splitting method could achieve higher throughput and optimal load balance. The average completion time of different sized flows is reduced by 24% on average and by up to 46%.     

An Adaptive AP Selection Scheme Based on RSS for Enhancing Positioning Accuracy

In recent years, the indoor positioning technologies have been recognized as core technologies for realizing smart space, a ubiquitous society, context awareness, and various location-based services. There are several approaches for positioning with radio signals, but the received signal strength (RSS)-based technology is considered a promising scheme because of its simplicity and practicality in implementation. In this paper, the positioning performance of the RSS value-based scheme is analyzed with respect to the location of access points (APs) and the number of APs in an indoor environment. An adaptive AP selection scheme and a base AP changing scheme are then proposed to enhance the positioning accuracy. In order to estimate the RSS characteristics, RSS values are measured as the distance between the AP and the receiver increases. The positioning performance is evaluated with differing AP numbers, which form a triangle or a quadrilateral. The performance of the proposed schemes is evaluated via experiments using wireless local area network APs. Results show that the performance of proposed schemes is enhanced compared to that of conventional scheme.     

General Architecture for MPEG-2/H.263/H.264/AVC to H.264/AVC Intra Frame Transcoding

The latest international video-coding standard H.264/AVC significantly achieves better coding performance compared to prior video coding standards such as MPEG-2 and H.263, which have been widely used in today’s digital video applications. To provide the interoperability between different coding standards, this paper proposes an efficient architecture for MPEG-2/H.263/H.264/AVC to H.264/AVC intra frame transcoding, using the original information such as discrete cosine transform (DCT) coefficients and coded mode type. Low-frequency components of DCT coefficients and a novel rate distortion cost function are used to select a set of candidate modes for rate distortion optimization (RDO) decision. For H.263 and H.264/AVC, a mode refinement scheme is utilized to eliminate unlikely modes before RDO mode decision, based on coded mode information. The experimental results, conducted on JM12.2 with fast C8MB mode decision, reveal that average 58%, 59% and 60% of computation (re-encoding) time can be saved for MPEG-2, H.263, H.264/AVC to H.264/AVC intra frame transcodings respectively, while preserving good coding performance when compared with complex cascaded pixel domain transcoding (CCPDT); or average 88% (a speed up factor of 8) when compared with CCPDT without considering fast C8MB. The proposed algorithm for H.264/AVC homogeneous transcoding is also compared to the simple cascaded pixel domain transcoding (with original mode reuse). The results of this comparison indicate that the proposed algorithm significantly outperforms the mode reuse algorithm in coding performance, with only slightly higher computation.     

Rapid acquisition of PN sequences with a new decision logic

In this paper, a fast acquisition scheme of pseudonoise (PN) sequences for direct-sequence spread-spectrum systems is proposed. The scheme exploits a new decision logic not only to estimate chip values, but also to check the reliability of the chip estimates. As a result, the internal state of a PN sequence generator is estimated more accurately. We derive the probability of finding a correct state estimate and show that the proposed scheme can reduce the average number of chips for acquisition when compared with the conventional scheme. It is also shown that the performance improvement is more noticeable in the moderate signal-to-noise-ratio range.     

Performance analysis of adaptive interleaving for OFDM systems

We proposed a novel interleaving technique for orthogonal frequency division multiplexing (OFDM), namely adaptive interleaving, which can break the bursty channel errors more effectively than traditional block interleaving. The technique rearranges the symbols according to instantaneous channel state information of the OFDM subcarriers so as to reduce or minimize the bit error rate (BER) of each OFDM frame. It is well suited to OFDM systems because the channel state information (CSI) values of the whole frame could be estimated at once when transmitted symbols are framed in the frequency dimension. Extensive simulations show that the proposed scheme can greatly improve the performance of the coded modulation systems utilizing block interleaving. Furthermore, we show that the adaptive interleaving out performs any other static interleaving schemes, even in the fast fading channel (with independent fading between symbols). We derived a semi-analytical bound for the BER of the adaptive interleaving scheme under correlated Rayleigh fading channels. Furthermore, we discussed the transmitter-receiver (interleaving pattern) synchronization problem     

一种集群计算并行I/O中启发式文件分类分配策略研究

提出了一种新的应用于集群计算环境的文件分配算法,即启发式文件分类分配算法(Heuristic FileSorted Assignment algorithm,HFSA),它在保证系统负载基本均衡的前提下,按相似的访问服务时间对每个待分配的数据文件进行磁盘分配.具体实现时,首先对待分配文件按它们的服务时间大小进行排序,然后从集群中选择一个结点的磁盘,将有序的若干文件连续地分配到该磁盘上,直接该磁盘达到最大负载时为止.通过对新策略与已有的贪婪文件分配法进行实验比较,结果表明:在访问响应时间方面,系统轻负载时,缩短了20.2%,系统重负载时,缩短了31.6%,并且数据访问的速率越高,由启发式文件分类分配策略所改善的性能就越明显.     

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%.     

Novel Rate Control Scheme for Low Delay Video Coding of HEVC

In this paper, a novel rate control scheme for low delay video coding of High Efficiency Video Coding (HEVC) is proposed. The proposed scheme is developed by considering a new temporal prediction structure of HEVC. In the proposed scheme, the relationship between bit rate and quantization step is exploited firstly to formulate an accurate quadratic rate‐quantization (R‐Q) model. Secondly, a method of determining the quantization parameters (QPs) for the first frames within a group of pictures is proposed. Thirdly, an accurate frame‐level bit allocation method is proposed for HEVC. Finally, based on the proposed R‐Q model and the target bit allocated for the frame, the QPs are predicted for coding tree units by using rate‐distortion (R‐D) optimization. We compare our scheme against that of three other state‐of‐the‐art rate control schemes. Experimental results show that the proposed rate control scheme can increase the Bjøntegaard delta peak signal‐to‐noise ratio by 0.65 dB and 0.09 dB on average compared with the JCTVC‐I0094 and JCTVC‐M0036 schemes, respectively, both of which have been implemented in an HEVC test model encoder; furthermore, the proposed scheme achieves a similar R‐D performance to Wang's scheme, as well as obtaining the smallest bit rate mismatch error of all the schemes.