Payload Length and Rate Adaptation for Multimedia Communications in Wireless LANs

We provide a theoretical framework for cross-layer design in multimedia communications to optimize single-user throughput by selecting the transmitted bit rate and payload size as a function of channel conditions for both additive white Gaussian noise (AWGN) and Nakagami-m fading channels. Numerical results reveal that careful payload length adaptation significantly improves the throughput performance at low signal to noise ratios (SNRs), while at higher SNRs, rate adaptation with higher payload lengths provides better throughput performance. Since we are interested in multimedia applications, we do not allow retransmissions in order to minimize latency and to reduce congestion on the wireless link and we assume that packet loss concealment will be used to compensate for lost packets. We also investigate the throughput and packet error rate performance over multipath frequency selective fading channels for typical payload sizes used in voice and video applications. We explore the difference in link adaptation thresholds for these payload sizes using the Nafteli Chayat multipath fading channel model, and we present a link adaptation scheme to maximize the throughput subject to a packet error rate constraint.     

Link Adaptation Using Dynamically Allocated Thresholds and Power Control

Link adaptation technique, in which the modulation and coding used in a communication system is changed as per the channel conditions is a very well investigated topic for link throughput maximization with widespread application in wireless access networks. Most of the known algorithms dynamically adjust transmitter data rate by comparing instantaneous SNR with pre-defined SNR switching thresholds, in order to maximize throughput while maintaining the desired quality of service. However, the use of incorrect or stale values of these pre-defined switching thresholds often leads to selection of erroneous modulation and coding schemes resulting in unsatisfactory throughput or quality of service. This work introduces a novel scheme which achieves the maximum possible throughput while maintaining the target quality of service by dynamically acquiring the threshold values of different modulation and coding schemes used in the system for a given value of block error rate based on measurement at the receiver. This helps in keeping the threshold look up table up to date, so that proper threshold values for mode switching is present for all channel conditions. Also, a relationship between the throughput and the accuracy of the threshold value calculation is provided so that these can be optimized depending on the user requirements. The performance evaluation shows that the proposed system outperforms the conventional link adaptation in various operating scenarios where pre-determined look up tables are not available.

     

Adaptive modulation and MIMO coding for broadband wireless datanetworks

Link adaptation techniques, where the modulation, coding rate, and/or other signal transmission parameters are dynamically adapted to the changing channel conditions, have emerged as powerful tools for increasing the data rate and spectral efficiency of wireless data-centric networks. While there has been significant progress on understanding the theoretical aspects of time adaptation in LA protocols, new challenges surface when dynamic transmission techniques are employed in broadband wireless networks with multiple signaling dimensions. Those additional dimensions are mainly frequency, especially in multicarrier systems, and space in multiple-antenna systems, particularly multiarray multiple-input multiple-output communication systems. We give an overview of the challenges and promises of link adaptation in future broadband wireless networks. We suggest guidelines to help in the design of robust, complexity/cost-effective algorithms for these future wireless networks     

Joint source-channel coding and power adaptation for energy efficient wireless video communications

We consider efficiently transmitting video over a hybrid wireless/wire-line network by optimally allocating resources across multiple protocol layers. Specifically, we present a framework of joint source-channel coding and power adaptation, where error resilient source coding, channel coding, and transmission power adaptation are jointly designed to optimize video quality given constraints on the total transmission energy and delay for each video frame. In particular, we consider the combination of two types of channel coding—inter-packet coding (at the transport layer) to provide protection against packet dropping in the wire-line network and intra-packet coding (at the link layer) to provide protection against bit errors in the wireless link. In both cases, we allow the coding rate to be adaptive to provide unequal error protection at both the packet and frame level. In addition to both types of channel coding, we also compensate for channel errors by adapting the transmission power used to send each packet. An efficient algorithm based on Lagrangian relaxation and the method of alternating variables is proposed to solve the resulting optimization problem. Simulation results are shown to illustrate the advantages of joint optimization across multiple layers.     

基于实数分类器的MIMO系统链路自适应

多输入多输出(Multiple-Input Multiple Output,MIMO)技术已被广泛应用于无线通信系统。由于信道中均衡、预编码、空间模式、调制和编码方案(Modulation and Coding Scheme,MCS)之间耦合的复杂性,在MIMO系统中执行链路自适应仍十分困难。通过实数分类器提出MIMO系统中的链路自适应方案来最大化频谱效率,并保持传输的可靠性。从信道状态信息(Channel State Information,CSI)中提取特征,并结合特定的代价函数选择调制和编码方案,基于最大化频谱效率的目标来选择空间模式。     

Forward high-speed wireless packet data service in IS-2000 - 1/spl times/EV-DV

This article describes enhancements to the physical layer design of single-carrier (1/spl times/) cdma2000/sup /spl reg// wireless networks introduced by Revision C of the IS-2000 standard (commonly referred to as 1/spl times/EV-DV). These improvements are mainly for the support of high-speed wireless packet data service on the forward (base station to mobile station) link. A new shared channel, the forward packet data channel, F-PDCH, is introduced to significantly increase spectral efficiency. This is accomplished through smart, channel-dependent scheduling of base station, BS, resources by taking advantage of the delay-tolerant nature of data traffic. This shared channel utilizes fast link adaptation through adaptive modulation and coding supported by the feedback of channel quality information. A form of hybrid ARQ type II technique commonly referred to as asynchronous adaptive incremental redundancy compensates for frame error events due to imperfect link adaptation. The impact of the new channel on voice capacity of the system is designed to be minimal because the F-PDCH utilizes resources that are not consumed by real-time users.     

A data link control protocol for broadband wireless networks with adaptive coding rate

One of the challenges of broadband wireless multimedia networks is to design a wireless data link control (W‐DLC) protocol which provides reliable data delivery over the noisy wireless link in an efficient manner. This paper presents our proposal of such a W‐DLC protocol, based on a new coding scheme called multi‐dimensional concatenated zigzag code. Both real‐time and non‐real‐time traffic can be efficiently supported by adopting different error control mechanisms. A distinct feature of the protocol is that the coding rate can be adapted to a wide range of channel conditions. Hence, our protocol utilizes the scarce wireless channel resources more efficiently. Copyright © 2005 John Wiley & Sons, Ltd.     

基于无线信道预测的自适应调制

该文研究了在平坦Rayleigh衰落信道下,借助于无线信道预测对一个基于非编码的MQAM自适应调制进行优化设计的问题。主要是通过采用无偏二阶估计的方法去研究一个已知的预测误差方差对最优化传输特性的影响。推导出基于预测的信噪比和预测误差方差的数据速率的最优解,以此对数据速率进行调整,使得频谱效率在误比特率的约束条件下达到最大化。通过仿真给出的数值解表明它能更好地将链路自适应地快速调整到信道的真实条件,提高整个系统的性能。     

Channel-quality-based opportunistic scheduling with ARQ in multi-rate wireless networks: modeling and analysis

In this paper, we develop a novel framework for analyzing radio link level performance for opportunistic scheduling with automatic repeat request (ARQ)-based error control in multi-rate wireless networks. The multi-rate transmission is assumed to be achieved through adaptive modulation and coding (AMC) to adjust the transmission rate according to the channel condition. The residual error effect due to each AMC setting is counteracted by means of a limited persistence ARQ protocol. The novelty of the proposed analytical framework lies in the fact that we are able to derive complete statistics (in terms of probability mass function) for both short-term and long-term performance measures such as system throughput, per-flow throughput, inter-success delay under both uncorrelated and correlated wireless channels. These performance measures can also be obtained in case of non-identical channels for different users. Analytical results are validated through simulations and the impacts of channel behavior on the different radio link level performance metrics are investigated.     

自适应调制编码技术及其在移动通信中的应用

自适应调制编码技术是克服无线信道的时变性的一种重要链路适应技术。常用的链路自适应技术有自适应功率控制、自适应调制编码、自适应帧长等。本文主要介绍了两种自适应调制编码技术，并分析了各自的优缺点。     

Combined turbo coding and unitary space-time modulation

Space-time coding is well understood for high data rate communications over wireless channels with perfect channel state information. On the other hand, channel coding for multiple transmit antennas when channel state information is unknown has only received limited attention. A new signaling scheme, named unitary space-time modulation, has been proposed for the latter case. In this paper, we consider the use of turbo coding together with unitary space-time modulation. We demonstrate that turbo coded space-time modulation systems are well suited to wireless communication systems when there is no channel state information, in the sense that the turbo coding improves the bit error rate (BER) performance of the system considerably. In particular, we observe that the turbo-coded system provides 10-15 dB coding gain at a BER of 10/sup -5/ compared to the unitary space-time modulation for various transmit and receive antenna diversity cases.     

光通信中基于脉冲调制的联合编码调制研究

为了研究无线光通信中数字脉冲调制与信道编码的联合编码调制问题,在给出调制时域结构的基础上,采用理论推导和数值仿真相结合的方法,分析了基于二进制线性分组码的开关键控、数字脉冲位置调制和数字脉冲间隔调制系统的差错性能,得出了信道编码纠错能力与调制阶数的结合关系。结果表明,采用开关键控调制的系统,其汉明距离与欧式距离等价,故不存在调制与信道编码参量匹配问题;脉冲位置调制系统在纠错个数大于调制阶数时,可取得较明显的编码调制增益,当纠错个数能被调制阶数整除时,编码和调制达到匹配;由于数字脉冲间隔调制的符号时隙不固定,存在错误传播现象,不宜与差错编码联合使用。所得结果可为无线光通信链路的设计提供参考。     

Performance analysis of variable rate: symbol-by-symbol adaptivebit interleaved coded modulation for Rayleigh fading channels

A challenging problem for a high-speed data service over wireless is to protect data over the error-prone fading channel in an effective way (high-bandwidth efficiency). We propose a bandwidth-efficient error correction scheme, namely variable rate adaptive bit interleaved coded modulation (ABICM), for the wireless mobile channel. The code rate and modulation level are varied according to the current channel state to exploit the time-varying nature of the wireless channel. Design challenges to achieve symbol-by-symbol adaptation are addressed. In particular, we address the criteria for choosing the family of component codes for the ABICM system. We propose a multilevel puncturing scheme that solves the problem of symbol-by-symbol adaptive puncturing and interleaving. The equivalent distance spectrum for variable rate symbol-by-symbol adaptive codes is introduced and analytical bounds on adaptive codes are derived that enable us to determine the optimal adaptation thresholds. Two operation modes, namely constant throughput and constant bit error rate (BER) controls, are proposed. It is found that there are significant gains relative to fixed-rate coding in terms of signal-to-noise ratio (SNR) and throughput. It is also found that the ABICM scheme is essentially not degraded when used with small interleaving depths. This makes the ABICM very suitable for real-time applications     

Delay constrained throughput optimisation with imperfect CSI using discrete adaptive power

Link adaptation is an effective tool to overcome fading effects in wireless links. However, time-varying adaptive transmission rate leads to queueing delay, and moreover, in practise, imperfect channel state information (CSI) is available for the transmitter to adapt its transmission rate and power. This article aims to consider practical constraints to enhance the link adaptation scheme. We reformulate and optimise buffer delay constrained throughput of a wireless link based on outdated noisy CSI. Discrete power adaptation is proposed, in which a limited number of feedback bits (just the index of transmission power level) is required, while the performance is improved compared to the constant power and is close to continuous adaptive power. A unified scheme is set-up, where constant, discrete or continuous adaptive power transmission is utilised considering average or instantaneous bit error rate constraints based on imperfect CSI. The effectiveness of our designs is evaluated by numerical evaluations.     

Two dimensional cross-layer optimization for packet transmission over fading channel

In this paper a single-input-single-output wireless data transmission system with adaptive modulation and coding over correlated fading channel is considered, where run-time power adjustment is not available. Higher layer data packets are enqueued into a finite size buffer space before being released into the time-varying wireless channel. Without fixing the physical layer error probability, the objective is to minimize the average joint packet loss rate due to both erroneous transmission and buffer overflow. Two optimization techniques are incorporated to achieve the best solution. The first is policy domain optimization that formulates the data rate adaptation design as classical Markov decision problem. The second is channel domain optimization that appropriately partitions the channel variation based on particular fading environment and carried traffic pattern. The derived policy domain analytical model can precisely map any policy design into various QoS performance metrics with finite buffer setup. We then propose a tractable suboptimization framework to produce different two-dimensional suboptimal solutions with scalable complexity-optimality tradeoff for practical implementations.     

Link Adaptation Based on Explicit Channel Estimation for Throughput Maximization Over Rayleigh Fading Channels

Link Adaptation (LA) has been proposed in the literature as a means of increasing the throughput obtained by each user in wireless communication systems. Due to its significant simplicity, LA has even been adopted as the standard adaptive coding technique for the next generation mobile communication system known as EDGE. In this paper, we present a novel link adaptation technique, valid for use in the case of Rayleigh fading wireless channels. The new technique is based on the blind channel estimation algorithm usually used for combating inter-symbol interference (ISI). The algorithm is presented and its efficiency in providing the maximum available throughput is illustrated by means of computer simulations.     

Link Adaptation Based on Explicit Channel Estimation for Throughput Maximization Over Rayleigh Fading Channels

Link Adaptation (LA) has been proposed in the literature as a means of increasing the throughput obtained by each user in wireless communication systems. Due to its significant simplicity, LA has even been adopted as the standard adaptive coding technique for the next generation mobile communication system known as EDGE. In this paper, we present a novel link adaptation technique, valid for use in the case of Rayleigh fading wireless channels. The new technique is based on the blind channel estimation algorithm usually used for combating inter-symbol interference (ISI). The algorithm is presented and its efficiency in providing the maximum available throughput is illustrated by means of computer simulations.     

A Novel Analytical Framework for Integrated Cross-Layer Study of Call-Level and Packet-Level QoS in Wireless Mobile Multimedia Networks

We present a novel integrated analytical framework for analyzing the quality-of-service (QoS) performance measures in a wireless mobile multimedia network. The framework integrates physical, radio link, and network layer parameters and protocols to analyze the call-level and packet-level performances. In the network layer, call admission control (CAC) is responsible for deciding whether an incoming call can be accepted or not so that the performances of the ongoing calls do not deteriorate below the acceptable level. Also, an adaptive channel allocation (ACA) scheme is used to maximize the utilization of the radio resources. In the data link layer, queue management and error control are used for non-real-time loss-sensitive traffic. In the physical layer, a finite state Markov channel (FSMC) is used to model channel fading, and adaptive modulation is used for rate adaptation according to channel quality. Various call-level and packet-level QoS measures for real-time, non-real-time, and best-effort traffic are obtained. The analytical results are validated by extensive simulations. Examples of the applications of the presented analytical framework are also provided     

Novel link adaptation for TETRA cellular systems

One of the main challenges for future wireless systems is to enhance the effective data throughput by exploiting the allocated bandwidth as much as possible. Among several approaches at different layers, one of the most important is constituted by the so‐called link adaptation (LA) techniques. They are characterized by the adaptation of a set of transmission parameters to the channel state in order to improve performance. In this context, this paper is focused on the analysis of a particular class of LA techniques called adaptive modulation and coding, where the modulation and coding rate of transmission can vary according to the channel behavior. In particular, a novel LA algorithm, namely the timed window (TW) method, suitable for time‐division Duplex systems is proposed here. The performance of the TW algorithm is evaluated by taking actual user mobility conditions, communication channel behavior, as well as the physical layer effects into account. Finally, it is important to stress that, even if the wireless bearer considered in this study is TETRA (TErrestrial Trunked RAdio), the approach is quite general and it can be of interest for other wireless networks and can be optimized for different channel models (e.g. TU50, HT200, etc.). Copyright © 2008 John Wiley & Sons, Ltd.     

Integrated scheduling and link adaptation for heterogeneous networks: design and performance analysis

ABSTRACT

This paper presents novel integrated scheduling and link adaptation (ISLA) schemes for links which have a common spectrum and possibly different rates and reliability constraints. We consider the problem of weighted sum average rate maximisation of wireless links. In an orthogonal transmission system, a link is selected using instantaneous signal-to-noise (SNR) of all the links, and its rate and power are set in a jointly optimised manner. To demonstrate the effectiveness of the proposed approach, we analyse ISLA schemes with continuous rate adaptation using constant or adaptive transmission powers. ISLA schemes are designed in similar settings for practical discrete rate-adaptive systems using adaptive modulation and coding. To this end, a design framework is presented, which relies on a partitioning of space of links’ SNRs into regions, similar to structured vector quantisation. Within this framework, a particular ISLA design with high performance and polynomial complexity is suggested. The proposed schemes are designed before the start of the transmission, based on closed-form solutions, and can easily be applied to multiple access or broadcast networks. We also analyse several schemes for comparison, which employ modified Round-Robin, opportunistic scheduling or online generalised scheduling. Numerical results demonstrate how the proposed ISLA schemes outperform the benchmark schemes and effectively meet various user requirements.¹