New Publications are available for Communication system theory
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New Publications are available now online for this publication.
Please follow the links to view the publication.Limited feedback scheme in the presence of feedback delay using Kalman filter
http://dl-live.theiet.org/content/conferences/10.1049/cp.2011.0669
In this paper, a Best-M feedback scheme using Kalman filter is proposed to reduce the uplink feedback overhead in the presence of feedback delay. Best-M feedback scheme is an effective method to reduce the number of the channel quality information (CQI) feedback. However, the CQI feedback after the conventional Best-M decision is inaccurate for base station (BS) scheduling due to the feedback delay and the time varying nature of the channel. A modified limited feedback scheme is proposed to use a Kalman filter to compensate for the performance degradation caused by the feedback delay in Best-M feedback scheme. In the proposed scheme, the user equipment (UE) exploits a Kalman filter to predict the signal to interference plus noise ratio (SINR). Then, the UE makes the Best-M decision based on this predicted SINR and sends the corresponding CQI back to the BS. The simulation results show that the Best-M feedback scheme using Kalman filter, which achieves the same feedback reduction as the conventional scheme, improves the system spectral efficiency.Tailoring optical chaos for communication and beyond
http://dl-live.theiet.org/content/conferences/10.1049/cp.2011.1321
Optical chaos tailoring by a fiber Bragg grating that introduces dispersive optical feedback into a semiconductor laser is demonstrated numerically. The results have potentials in improving chaotic communication and chaotic ranging. (2 pages)A hybrid low complexity decoding of LDPC codes
http://dl-live.theiet.org/content/conferences/10.1049/cp.2010.0630
An efficient early stopping criterion based on the evolution of the number of reliable variable nodes (NRVN) is proposed for low density parity check (LDPC) codes in this paper. This criterion can significantly reduce the average number of iterations (ANI) at low to middle signal to noise ratios (SNRs) with minimal performance degradation. Furthermore, a hybrid decoding method is presented to reduce the decoding complexity of LDPC codes by combining the proposed stopping criterion and forced convergence (FC) decoding algorithm. Compared with the classic belief propagation (BP) decoding, significant computation complexity reduction can be achieved with small computation overhead.Optical-layer multicast based on network coding (Abstract only)
http://dl-live.theiet.org/content/conferences/10.1049/cp.2010.0708
Summary form only given. Optical-layer multicast will be a promising solution for high-speed and high-bandwidth multicast applications. While network coding can achieve maximum multicast rate by allowing intermediate network nodes to combine the data received from different incoming links. Therefore, how to apply network coding to optical-layer multicast has been attracting much attention. Due to the limitation of optical buffering and optical processing capabilities, several critical issues should be addressed, e.g., constructing an optical multicast tree with minimum coding resource and finding a network coding scheme realized by practical optical hardware. In this talk, we introduce our research and development activities on the optical-layer multicast based on networking coding, which is partly supported by NSFC (No. 60772024). Firstly, we discuss an improved quantum-inspired evolutionary algorithm that allows different chromosomes to select different updating parameter at each generation. Its efficiency is verified with high success ratio and small number of coding nodes in constructing a multicast tree. Then, we present a network coding scheme that recoveries the original signals at each coding node. The kernel function can be realized by logical bit shifting and bitwise XOR in optical domain. Finally, we build an experimental network, on which optical multicast via network coding is successfully demonstrated.Research and implementation of IPv6 controllable multicast in the campus network
http://dl-live.theiet.org/content/conferences/10.1049/cp.2010.0615
In allusion to PIM-SSM multicast model work mechanism in the IPv6 campus network, analyzing key technologies of controllable multicast, making use of controllability of network devices for multicast reported packets of multicast user joining (S,G) channel, putting forward PIMSSM controllable multicast realization framework in the IPv6 campus network. Combining with deployment of multicast management server, multicast authentication server, multicast security policy server and multicast monitoring server, constituting management and control with multicast address, multicast source, multicast receiver, and real time monitoring to multicast operating status, achieving muticast operation management and control in order in the IPv6 campus network.User-oriented cross-layer architecture in cognitive networks
http://dl-live.theiet.org/content/conferences/10.1049/cp.2010.0741
Research of the cognitive network is a hotspot problem in telecommunication field. In this paper, we proposed user-orientated cross-layer architecture in cognitive network where the user's experience is with great consideration. In order to guarantee the use's experience of the service, we propose a novel model to evaluate the users' quality of experience (QoE) which is related to the parameters of quality of service (QoS) and the users' behavior model. Since the network is user-oriented, different users should have the distinguishing cross-layer algorithms and the cross-layer algorithm should also be dynamic to adapt the diverse environment. So we introduce the concept of dynamic cross-layer design to this architecture in order to enable a dynamic adjustment to the cross-layer algorithm according to the users' QoE.Re-keying scheme for secure multicast based on multi-group key tree
http://dl-live.theiet.org/content/conferences/10.1049/cp.2010.0722
A re-keying scheme with multi-group key tree shared between multicast groups under group members (GMs) overlapping condition was presented in this paper, and solved the problem that re-keying cost of multicast groups had a linear relationship with the number of multicast groups. In this scheme, leaf nodes on tree were private keys of GMs, extended root nodes were group keys and keys corresponding to all nodes were updated using Pseudo. Random Function (PRF) and Exclusive OR (XOR) operation. The simulation results of verification system show that this scheme obviously improves re-keying performance to some extent under communication scene with multicast groups.A reduced complexity SNR estimator
http://dl-live.theiet.org/content/conferences/10.1049/cp.2009.1237
This paper presents a novel reduced complexity SNR estimator, whose performances are comparable with the maximum-likelihood estimator while saving one half of the floating point multiplications and one third of the floating point sums. A detailed statistical description of the estimate, as well as simulation results, are provided. (4 pages)Cognitive radio based spectrum assignment for heterogeneous multicast terrestrial communication systems with different transmission rate requirements
http://dl-live.theiet.org/content/conferences/10.1049/ic_20080402
This paper investigates two channel assignment schemes that select channels based on optimizing the coverage area supported by a multicasting terrestrial network. It considers the interaction of two different channel assignment schemes -Channel Priority and Maximal Difference when used together. It is found that it is good to combine different schemes together because the mixed schemes can exploit the best features of each individual scheme. When applying different transmission rates in each scheme, the interaction still remains positive. In these circumstances the Channel Priority scheme will automatically select additional channels to cope with the increased SINR threshold. (5 pages)An adaptive playout algorithm based on kernel estimation
http://dl-live.theiet.org/content/conferences/10.1049/cp_20081028
Adaptive playout algorithms play a very important role in guaranteeing the QoS (quality of service) in VoIP (voice over IP). The prediction of network delay in adaptive playout algorithms is essentially a time series analysis problem. Currently most delay prediction methods are based on parameter estimation. However, the strong parameter assumptions in parameter estimation usually could not correspond to the actual situation, which leads to poor performances. This paper proposes a new adaptive playout algorithm based on kernel estimation which is a type of non- parameter estimation. Simulation results show that the algorithm can adapt to the changeful network environment and reduce the loss and delay significantly.Design of input signal waveshapes for ISI free transmission in bandlimited channels
http://dl-live.theiet.org/content/conferences/10.1049/ic_20070698
In this paper, we propose a new technique to eliminate ISI in bandlimited channels, whose characteristics are known a priori, by designing channel-specific input and output waveshapes that can be constrained to the same time-duration. The problem has been worked out for two cases where the channel can be modelled as (i) rational transfer function and (ii) distributed parameter model. For both cases, there is zero ISI at the receiver input and hence only a matched filter is required to maximize the peak signal to noise ratio. For channels as in case (i), the lengths of the associated input and output waveshapes can be made arbitrarily small, thus permitting high signalling rates. For channels as in case (ii), the signalling rate works out to be a function of the line parameters and the line length. Performance analysis, in the presence of channel noise, shows that the SNR, the input power requirements as well as the tolerance to timing offsets, for a defined performance index, is dependent on the signal waveshape.Multi-user UWB-IR systems with interleaved coding-modulation on multipath fading channels
http://dl-live.theiet.org/content/conferences/10.1049/ic_20060032
Interleaved coding-modulation (ICM) is a recently proposed method for ultra-wideband impulse radio (UWB-IR) systems. ICM exploits the concept of chip interleaving, allowing to alleviate the problem of inter-symbol and inter-pulse interference commonly present in high data rate UWB-IR systems. In this paper, the authors extend previous work on ICM to scarcely populated multi-user scenarios. The authors propose a design of a deterministic chip interleaver based on time-hopping hyperbolic congruence sequences. The authors also review the main parameters of the random and hyperbolic interleavers. Our results indicate that the proposed type of interleaver yields similar performance to random interleavers but with the advantage of simpler implementation.Range resolution improvement in passive coherent location radar systems using multiple FM radio channels
http://dl-live.theiet.org/content/conferences/10.1049/ic_20060025
Passive coherent location (PCL) radar systems that use single FM radio channel signal as illuminator of opportunity have limited range resolution due to low modulation bandwidth and high dependence on the content that is being broadcasted from the FM station. An improvement in range resolution is obtained by using multiple adjacent FM channels, emitted from co-sited transmitters, which is often the case in large towns in countries, where the FM channel allocations are relatively weakly regulated. The proposed scheme computes the autocorrelation function of the signal directly received from the FM co-located transmitter, and compares it to the cross-ambiguity function, obtained from direct and target scattered signals. The geometry of the problem is like in the case of monostatic radar. The range information is obtained by the delay between the cross-ambiguity function and the autocorrelation function. It is shown that down to -37dB signal to noise ratio (SNR) the autocorrelation function of 7 FM channels with different contents can be successfully extracted from the cross-ambiguity function. The detection of the time delays is a linear estimation problem. The issue of time-delay estimation is a known topic of research. A powerful estimator can be found.Performance analysis of antipodal CSK communication system in a noisy multiuser environment
http://dl-live.theiet.org/content/conferences/10.1049/cp_20061321
To provide a quantitative description for noisy multiuser performance of digital communications with chaos, this paper presents an effective approach to analyzing the noisy multiuser performance of an antipodal chaos-shift-keying system with a coherent correlator. Using a logistic map as the discrete chaos generator, the BER is derived in terms of the signal-to-noise ratios, the number of users and the length of chaotic sequence. The BER and its upon bound for single-user systems are the special cases of the multiuser system. The calculated BERs are consistent with those found from simulations. (3 pages)Optimization flow control with DFP variable metric method algorithm
http://dl-live.theiet.org/content/conferences/10.1049/cp_20061549
In this paper, we proposed a distributed algorithm for obtaining the optimal fair bandwidth allocation based on local algorithms and measurements. The approach we use is draw from Davidon-Fletcher-Powell variable metric method (so called quasi-Newton method) in optimization theory, and the simulation result show that the algorithm converge faster than other algorithm used before. (3 pages)Implementation of local re-broadcasting and LAN services over an Ethernet passive optical network
http://dl-live.theiet.org/content/conferences/10.1049/cp_20050598
A novel technique to implement value-added services over an Ethernet passive optical network is proposed and optical link performances of 155-Mb/s LAN, 50-Mb/s digital TV re-broadcasting, and 750-Mb/s downstream delivery are experimentally evaluated. (2 pages)Field demonstrator of a chaos encrypted optical transmission system
http://dl-live.theiet.org/content/conferences/10.1049/cp_20050780
We report on the successful demonstration of a chaos encrypted gigabit optical transmission system utilizing a commercially available optical network installed in the metropolitan area of Athens. (2 pages)Delivering satellite telecommunications to the Ministry of Defence, where quality of service can mean the difference between life and death
http://dl-live.theiet.org/content/conferences/10.1049/ic_20040038
Satellite telecommunications services for military and government customers are required to provide reliable, flexible and secure communications, at short notice, anywhere in the world. The UK Armed Forces have come to rely increasingly on this capability. Under contract, Paradigm Services have taken over, from the UK Ministry of Defence, the provision and operation of MOD's satellite telecommunications network. Paradigm delivers managed services across the military satcom network and makes customer care provision. Paradigm has two types of customer. Firstly, there are the Ministry of Defence officials who pay the bill and require proof that the service delivered is providing good value for money. Secondly, there are the end users, some of whom may be sitting in comfortable offices, while others may be in environmentally and/or militarily hostile areas. In a commercial network, these people would be the traditional customers who rely on the integrity of the network and whose day to day activities are impacted by the delivery of the service. QoS for individual users is important. Unlike commercial telecommunications systems, however, an end user cannot terminate their contract. But, if we get the QoS wrong, it can be the difference between life and death for some people. This is a big motivator. The paper discusses the issues around QoS delivery, where the end user is not the customer, in the traditional sense, but is the person impacted by the failure to deliver a quality service.Intelligent packet scheduler for general packet radio service
http://dl-live.theiet.org/content/conferences/10.1049/ic_20040014
The general packet radio service (GPRS) augments GSM to provide packet switched data services to mobile users. Packet scheduling in GPRS is dynamic and several scheduling techniques have been implemented, for example round robin; however, these generally assure only best effort quality of service. We compare prominent scheduling algorithms by simulation of Web and E-mail traffic finding that earliest deadline first and first come first served scheduling perform well with few users, but round robin is preferable with large numbers of users. We introduce a novel scheduling algorithm, based on reinforcement learning, for scheduling packets according to quality of service. Simulation studies show that it outperforms a naive prioritised round robin algorithm and can adapt to changing network conditions.User quality of service perception in 3G mobile networks
http://dl-live.theiet.org/content/conferences/10.1049/ic_20040018
UMTS reflects the latest technical and standardisation efforts to realise and deploy a QoS-enabled mobile network infrastructure that can support QoS-dependent services and applications. The next challenge, therefore, is how to use and configure the network efficiently in a way which meets user expectation while conserving network resources. In order to achieve that, it is important to have qualitative information on how different QoS levels are perceived by the user. The network operator/service provider should be aware of the point at which the user. perceives the effect of QoS fluctuation, performing a reasoned and accurate dimensioning of the network. Many subjective studies have already been undertaken for conversational and streaming services, whereas we focus on the analysis of user QoS perception in the context of mobile, interactive services, specifically Internet access, Web browsing, and messaging.Quality of service to subscriber
http://dl-live.theiet.org/content/conferences/10.1049/ic_20040015
The mobile telecom industry has been able to specify what is required for monitoring and measuring quality of service (QoS) which means specifying the level of quality which should be provided for a service. By service, the paper refers to the mobile network's service level not service connotations like customer care, sales support, repair time, provisioning, etc. Here, QoS specifically refers to services that are like those offered by 3G networks, e.g., MMS, picture and video messaging, video streaming, video downloads, etc. Conventional packet services like E-mail and Web browsing are also referred to. It is forecast that, in about a year's time, vendors and operators of mobile networks will firmly establish the exact network parameters that are essential to classify and measure quality of service. These parameters, as we understand today, will be derived mostly from the IP environment; packet loss, maximum packet discard rate, reliability and throughput being some of them. The objective of the paper is to introduce and develop a QoS concept that caters to the customer's demand rather than the service demand. Examples are provided on how to set up such QoS profiles. The QoS profiles can be termed MyQoS, personalised QoS or QoSS (quality of service to subscriber).Defining and monitoring QOS metrics in the next generation wireless networks
http://dl-live.theiet.org/content/conferences/10.1049/ic_20040013
Defining and monitoring QoS metrics for service level agreements (SLA) in next generation wireless networks is a challenging task for a service provider. Some complex issues that need to be addressed are: different QoS requirements for different classes of applications; support of real-time applications (such as voice, video) over packet based networks; heterogeneous characteristics of wireless and wireline networks; mobility of users; evolving applications; services; etc. Existing SLAs and their monitoring methodologies and tools lack the sophistication required to address these new issues. This work provides a framework to address them. It classifies next generation applications into several classes and identifies the important QoS metrics for each class. It provides the measurement methodologies for these QoS metrics so that a service provider may establish and verify a service level agreements with his/her customers. It identifies the requirements for the tools in order to monitor/measure these new QoS metrics. Finally, it discusses the challenges faced by service providers while implementing these recommendations.Quality differentiation and economic efficiency
http://dl-live.theiet.org/content/conferences/10.1049/ic_20040010
Summary form only given. The capability to differentiate quality of service is important for networks in order to improve their economic efficiency. In such a 'market-managed' network, customers can express their preferences and acquire the resources for which they are willing to pay. Prices, if set correctly, can provide the right incentives so that network resources can be used more efficiently, and hence the competitive position of the network operator can be improved. Traditional approaches to providing QoS place the network at the centre of making decisions about how to customize and differentiate services. However, end-users (devices) may benefit greatly from the ability to 'construct' their own services. This is in line with the 'end-to-end principle', in which the internal network nodes are kept simple and complexity is moved to the edges where information about user utility resides. If users can obtain greater surpluses, then the network can probably obtain more revenue. The challenge is therefore to provide the edges with flexibility to express their preferences in terms of quality and payments, while enabling the network provider to control the pricing and the provisioning process. We discuss the above concepts and provide certain microeconomic arguments that support them.New QoS metrics and application layer proxy for GPRS/UMTS Internet access
http://dl-live.theiet.org/content/conferences/10.1049/ic_20040039
Despite the fact that the third generation (3G) mobile system is expected to provide a wide range of services, including data access, it has been designed primarily for voice services and it provides QoS guarantee in terms of bit rate and delay. These QoS metrics are more suitable for a real time streaming service rather than the dominant data service in the Internet. The paper proposes new QoS metrics, and response time, for Internet data access in the Universal Mobile Telecommunication System (UMTS). The new metrics reflect the user's expectation of QoS. Networks can change delay and bit rate according to the current situation while keeping the response time constant. A proxy based architecture is presented to support the new QoS guarantee. The proxy caches files and helps the wireless network to schedule file transfer according to the system capacity and user demand. Theoretical analysis and simulation show that the new scheme can improve user-level performance and network utilization.Soft handover performance for UMTS operations
http://dl-live.theiet.org/content/conferences/10.1049/cp_20030324
Assessment of handover performance is key for ensuring the desired quality of service to mobile communications. Handover algorithms in UMTS are quite sophisticated and handover performance depends on a range of control parameters that need to be tuned to accommodate the operating environments and contribute to achieving the target QoS objectives. The paper investigates the performance of soft handover for UMTS systems. To this end, the paper introduces a detailed model for handover handling and exercises the model under a range of combinations regarding operating environment and handover control parameters. The conclusions drawn are useful for identifying the range of control parameters enabling robust and effective operation of UMTS systems.Personal distributed environment securing the dynamic service platforms beyond 3G
http://dl-live.theiet.org/content/conferences/10.1049/cp_20030329
Future mobile systems are expected to offer users flexible access to information and services using a combination of different end-user devices in a personal distributed environment (PDE). With PDEs able to operate over multiple air interfaces and heterogeneous networks, requiring seamless and rapid service provision, a flexible and fair trading of communication services is required. For. this reason, a digital marketplace (DMP) is proposed. The DMP is based on an agent framework capable of enabling real-time service negotiation over disparate networks according to users' price and QoS requirements. The paper discusses the security threats and challenges to the PDE, and also to a DMP implementation.Enhanced trellis extracted synchronisation technique for practical implementation
http://dl-live.theiet.org/content/conferences/10.1049/ic_19990847
Intrinsic synchronisation algorithms increase the data throughput of a system at the expense of processing. The trellis extracted synchronisation technique (TEST) is an intrinsic synchronisation algorithm for the combined decoding and synchronisation of error control block codes. This paper illustrates two enhancements to the TEST algorithm which improves synchronisation performance and reduces processing. Inherent errors due to the linearity and cyclic properties of block codes are corrected. A simple estimate of future synchronisation points in a data stream can be efficiently used to decrease the processing with no expense of coding performance. (5 pages)Extracting accurate channel estimation from decoder metrics
http://dl-live.theiet.org/content/conferences/10.1049/ic_19990843
The requirement for accurate channel estimation exists for a number of communications systems. The method described in this paper was specifically designed for turbo code systems which must have knowledge of the channel to operate. However, this method is applicable to any system using a code that may be represented by a trellis structure. It purely digital solution that is ideally suited for implementation on a DSP device. (5 pages)Novel DSP-based adaptive synchronisers
http://dl-live.theiet.org/content/conferences/10.1049/ic_19990848
In a digital communication system, synchronisation between the transmitter and the receiver is essential to ensure reliable data transmission. There are three main categories of synchronisation associated with a typical data receiver system i.e.: (i) carrier synchronisation; (ii) symbol synchronisation; and (iii) frame synchronisation. The aims of the research are to acquire carrier, symbol and frame synchronisation with: (i) minimum transmission overheads (intrinsic synchroniser); (ii) minimum synchroniser complexity; and (iii) an adaptive synchronisation capability. This paper emphasises symbol synchronisation. Since one of the aims of the research is to acquire symbol synchronisation with minimum overheads, data derived synchronisation is emphasised here. Data derived synchronisation is a method of symbol synchronisation that allows timing information to be extracted from the message signal itself. There are three types of data derived symbol synchronisation algorithms available. The modulation derived synchronisation (MDS) is chosen as the basis of this paper. (7 pages)A comparison of blind channel estimation and equalisation techniques for a fading environment
http://dl-live.theiet.org/content/conferences/10.1049/cp_19980006
A number of blind channel estimation techniques has previously been presented. We consider the performance of blind techniques for a time-varying, fading, environment. The three techniques we investigate are Tong's (1994), subspace, and multiple survivor Viterbi methods due to their fast convergence and robustness to additive Gaussian noise. Their performance is compared with the non-blind cross-correlation scheme used in the GSM system. We also give the bit error rates (BER) in terms of the SNR after passing the resulting channel estimates from the three techniques to the Viterbi decoder. It is shown that both Tong's and the subspace methods outperform the multiple survivor Viterbi algorithm. Also, Tong's and subspace method show only a 5 dB loss in SNR as compared with the non-blind cross-correlation technique.Nonlinear dynamics and noise cancellation
http://dl-live.theiet.org/content/conferences/10.1049/ic_19971369
The use of linear and nonlinear prediction in forming the inverse of a linear system is explored. The topic is introduced from the perspective of communications channel equalisers, where the signal of interest is stochastic, and from the perspective of nonlinear noise cancelling, where the signal of interest may be deterministic and chaotic. In both applications a nonlinear inverse to a linear system can produce better results than a linear inverse. The nonlinear architectures considered are linear-in-the-parameter radial basis function (RBF) and Volterra series (VS) networks. The application of nonlinear filtering techniques to the cancellation of noise in a linear duct is also considered. It is demonstrated that the required inverse is provided by the parallel connection of a linear and nonlinear network of different memory lengths. (6 pages)Optimised MSE and delay for blind equalizers
http://dl-live.theiet.org/content/conferences/10.1049/ic_19971316
Multipath Rayleigh fading channels which have fast time variation can cause loss of tracking in equalizers, from which recovery without retraining is extremely difficult. In such situations, blind techniques which recover the optimum MSE over all possible symbol delays can be helpful. This paper explores one such technique. Simulation results demonstrate its suitability. (5 pages)Blind estimation of FIR communication channels via probabilistic methods
http://dl-live.theiet.org/content/conferences/10.1049/ic_19971314
In this paper a new blind estimation method for FIR channels is presented. The transmitted signal under consideration is multilevel PAM. The method uses the information contained in the conditional probability density functions of the received signal samples. Simulation results show that the method works well for low SNRs and short record lengths. The price to pay is the high complexity. (6 pages)Blind equalization - combinations of Bussgang- and higher-order-statistics-based methods
http://dl-live.theiet.org/content/conferences/10.1049/ic_19971318
A new generation of higher-order-statistics (HOS) based blind equalization techniques, like the eigenvector algorithm (EVA) published in 1997 by B. Jelonnek, D. Boss and K.-D. Kammeyer, leads to a much better performance. In this paper these techniques are used to initialize a conventionally Bussgang algorithm. The purpose is to show, through simulations, that combinations of both techniques have the ability to improve the equalization quality and to reduce the computational complexity as well. Also, we consider the modifications of these algorithms for the application of a fractionally-spaced (FTS) equalizer. (6 pages)An analytical approach to charging mechanisms using control theory
http://dl-live.theiet.org/content/conferences/10.1049/ic_19971105
We suggest how control theory may be used to model and analyse telecommunication network charging mechanisms. We propose a control theory model which was used to analyse the dynamic (also known as `smart market' or `free market') charging scheme. We show how the analytical model was formalised and analysed to determine the long term or steady state (SS), in control theory terminology, and short term or transient responses of the mechanism. We show how the system's performance varies as the parameters change. We also discuss the limitations of this method of analysis and in particular how this affects the results of the dynamic charging scheme analysis. (9 pages)Application of channel matched filters and direct coefficients calculation for low complexity high bit rate equalisation
http://dl-live.theiet.org/content/conferences/10.1049/ic_19960760
For high data rate mobile communication systems, because of the channel time-delay-spread, the equalisers are critical for combating inter-symbol-interference (ISI). The required time for equalisation training is one of the main limitations affecting the data rate and throughput. This study introduces a new method for equalisation using a channel matched filter (CMF) which combines a speed of coefficient calculation comparable with the simple least-mean-square (LMS) algorithm, with an equaliser filter performance which can exceed that of the recursive-least squares (RLS) algorithm. Synchronisation, channel tracking and frequency offset problems are also considered. (7 pages)A comparison of cyclostationary blind equalisation techniques
http://dl-live.theiet.org/content/conferences/10.1049/ic_19960761
Interest in blind equalisation has grown in recent years due to the fact that in some communication systems the transmission of a training sequence is not physically feasible. Most blind equalisation schemes to date have sampled the channel output at the symbol rate to produce a stationary output sequence. This paper presents a comparison of the performance of two algorithms for nonminimum phase blind channel equalisation using fractionally-spaced sampling in the receiver. The major problem associated with nonminimum phase channel equalisation is that the measured output statistics must preserve the phase characteristics of the channel. Cyclostationary statistics, unlike the conventional second-order statistics, have been shown to be efficient in this respect. Two families of algorithms are available which exploit the cyclostationary nature of the oversampled received signal in different ways. Both philosophies provide algorithms which are superior to existing symbol-spaced blind equalisation techniques in terms of reliability and speed of convergence. To date no comparison of these two approaches has been carried out and this paper examines these techniques by means of simulation using multipath channels. (6 pages)Intrinsic block, symbol and bit synchronisation
http://dl-live.theiet.org/content/conferences/10.1049/ic_19951401
The majority of the classical methods for synchronisation for digital communications systems require a hardware overhead and additional data added to the information stream to provide effective synchronisation. The paper introduces a novel synchronisation technique which utilises auxiliary information obtained from the soft maximum likelihood trellis decoding (SMLD) of the received signal, to obtain bit, symbol and block synchronisation. This combines the operations of the decoder, demodulator, and synchronisation system. (5 pages)Joint channel estimation and data detection using a blind Bayesian decision feedback equaliser
http://dl-live.theiet.org/content/conferences/10.1049/ic_19950899
A blind Bayesian decision feedback equaliser has been developed for joint channel estimation and symbol detection. It has been shown how the complete blind equaliser is built up with many identical adaptive units. Each of these units consists of a bank of simple least mean square channel estimators and Bayesian decision feedback equalisers. An efficient parallel implementation can therefore be realised readily. Simulation results have demonstrated fast convergence of this blind equaliser. Convergence can generally be achieved in less than 100 symbols when binary symbol constellation is used and within a few hundred symbols when a 4-level symbol constellation is used. (5 pages)An adaptive decision feedback equalizer with shaped training for mobile communication FM receivers
http://dl-live.theiet.org/content/conferences/10.1049/cp_19951144
The paper describes an adaptive equalizer suitable for digital wireless systems. These systems are normally defined to work in indoor environments where channels show low delay spread values. For this reason, the possibility of signal equalization is not usually taken into account. Moreover, simplicity and cost are important parameters for these systems, and therefore simple and inexpensive receivers are used whenever it is possible. Because of their high capacity and low cost, it is interesting to study new applications based on these systems, for example, outdoor applications. However, these applications show higher delay spread values are necessary to perform an equalization process to maintain the communication quality. The problem appears when a limiter/discriminator receiver is used in the system due to the fact that this produces different distortions over the received signal which make the equalization process difficult. On the other hand, the modulation used in these systems may sometimes show modulation index variations that must be compensated for. The paper shows an equalization method that can perform the equalization process over the received baseband signal, compensating all distortions mentioned by using a shaped training sequence.The acquisition performance of delay-lock loops in noise
http://dl-live.theiet.org/content/conferences/10.1049/cp_19951145
Describes results of a computer simulation of a re-configurable delay-lock loop (DLL) which allows fast acquisition of code synchronisation in very poor input SNR conditions. The DLL is based on an analogue prototype but is implemented digitally and this allows the loop to be switched very easily from a noncoherent configuration during initial acquisition to a coherent loop after code, carrier and data bit synchronisation have been achieved. Consequently, the jitter performance of the loop is significantly improved during the normal tracking phase. The loop can also be switched from a conventional 1Δ or 2Δ configuration to a much wider loop, such as a 4Δ loop to give improved tracking performance under conditions of very high code-rate Doppler frequency shifts found in low earth orbit satellite systems.Mean time to lose lock for a coherent extended tracking range delay-locked loop
http://dl-live.theiet.org/content/conferences/10.1049/cp_19951142
The delay-locked loop (DLL) is an appropriate device to guarantee fine synchronization for direct-sequence spread spectrum systems. The mean time to lose lock (MTLL) is a good performance criteria for the DLL operating at low signal to noise ratios. In the paper, the influence of the extended tracking range on the MTLL for a coherent DLL is investigated. A second order loop with passive loop filter is considered. The MTLL is approximated by an explicit expression for the leading order term based on results by Kramers (1940). The results compare the MTLL for different coherent DLL structures.Phase inference and error surface analysis of a blind non minimum phase channel equalizer
http://dl-live.theiet.org/content/conferences/10.1049/ic_19950897
Blind equalization of a non-minimum phase (NMP) channel can be achieved with an infinite impulse response (IIR) predictor and an all pass filter. The adaptive IIR predictor minimises the square of the prediction error with the simplified gradient least mean square algorithm which possesses the self-stabilizing property, and the all pass filter is adapted to minimise the fourth moment of its output. This paper analyses the error surface of the all pass filter output for an NMP channel and proposes a new scheme to initialise the all pass filter from the IIR predictor coefficients. A method used to achieve blind equalization with only second order statistics which has immunity to local minima is also proposed. Simulation results are used to support the work. (6 pages)Using method of slowly changing amplitudes for analysing oscillations in phase-locked loops
http://dl-live.theiet.org/content/conferences/10.1049/cp_19951143
The phase-locked loop has been found to be a very useful element in many types of communication systems. It is used in two fundamentally different ways [Gardner (1981), Encinas (1993)]: (1) For demodulation, where it is used to follow phase or frequency modulation of the input signal. (2) For tracking a carrier or synchronizing signal which may vary in frequency with time. The authors discuss stability loss criteria, loop oscillations and stationary loop oscillations.Extension of Dempster-Shafer theory and application to fault diagnosis in communication systems
http://dl-live.theiet.org/content/conferences/10.1049/cp_19940643
A novel method is presented for propagating uncertainty that also calculates measures of contradictions in the input data. This method can improve the performance of a reason maintenance system by ranking the contradictions and resolving the most severe of these first. An example shows the application of this technique to fault diagnosis in a communication system.Single-carrier frequency domain equalisation with hierarchical constellations: an efficient transmission technique for broadcast and multicast systems
http://dl-live.theiet.org/content/journals/10.1049/iet-com.2011.0674
Orthogonal frequency division multiplexing (OFDM) schemes are the choice modulation for broadband wireless broadcast and multicast systems. However, OFDM schemes have important limitations such as high-envelope fluctuation of the transmitted signals and its sensibility to carrier frequency errors. When these limitations prove critical the authors consider single-carrier frequency-domain equalisation (SC-FDE) schemes, that allow much higher power efficiency because of lower envelope fluctuations. The overall performance can be further improved if the conventional linear FDE is replaced by an iterative FDE such as an iterative block decision feedback equaliser (IB-DFE). Conventional IB-DFE are usually designed for a quadrature phase shift keying constellation. However, it is strongly recommended that broadcast and multicast systems employ hierarchical constellations with several classes of bits with different error protection. In this study the authors consider the use of SC-FDE schemes combined with IB-DFE receivers in broadband wireless broadcast and multicast systems. The authors emphasise the advantages of these schemes and the authors present IB-DFE designs suitable for hierarchical constellations with several classes of bits with different error protection.Multiple-input multiple-output ultra-wide band channel modelling method based on ray tracing
http://dl-live.theiet.org/content/journals/10.1049/iet-com.2011.0265
In this work, the authors have developed a deterministic Ultra Wide Band (UWB) channel model for indoor environment using both ray-tracing technique and the art of computer game technology in 3D Game Studio (game development tool). In the developed model, the characteristics of indoor environment such as texture, transparency etc. can be taken into consideration while indoor parameters such as room size, objects position etc. can be interactively changed. Each time, indoor environment is changed, the program is compiled and hence, the underlying ray-tracing captures the updated indoor environment. It is the key novelty of the authors’ developed model and it has been so incorporated to make the authors’ model independent of any fixed (pre-defined) indoor environment. The developed model is compared against the standard statistical UWB channel model based on certain parameters such as delay spread etc. to address its validity and accuracy. The model is then enhanced to use multiple antennas on both sides of the system and capture the channel response accordingly. Finally, the developed model has been tested over a range of frequencies to see frequency effect on the channel impulse response. The simulation results have been presented and discussed in the simulation section.Optimised priority assignment mechanism for biomedical wireless sensor networks
http://dl-live.theiet.org/content/journals/10.1049/iet-wss.2011.0061
Biomedical wireless sensor networks (BWSNs) need to guarantee reliable and timely transfer of crucial data in emergency and time critical situations. Current schemes use static/fixed priority assignment mechanism based on source of data and not on the urgency of the data in different situations. This study presents a novel data transmission scheme, called optimised priority assignment mechanism (OPAM) for BWSNs. The proposed work dynamically schedule different types of data flows based on their time critical nature. It smartly assigns priority to individual data packets rather to particular service or flow by continuously monitoring queuing delay providing guaranteed end-to-end quality of service (QoS) without invoking any congestion control and avoidance mechanism. Experimental results show that OPAM performs better in terms of average throughput and end-to-end delay 55 and 20.5%, respectively, than routing service framework for standard BWSNs.Competitive decoders for turbo-like chaos-based systems
http://dl-live.theiet.org/content/journals/10.1049/iet-com.2011.0865
Recent work has shown that chaos-based communication systems can yield performances as good as their non-chaotic counterparts in additive white Gaussian noise (AWGN), and behave even better in flat-fading channels. However, much of this work relies on computer simulations and there is still a need to study in depth the implementation issues of such systems. The authors address for the first time a fixed-point arithmetic implementation of the iterative decoding algorithm for a recently proposed and successful class of parallel concatenated chaos-based coded modulations. The novel digital signal processor (DSP) results demonstrate that it is possible to implement in standard hardware competitive chaos-based communication systems.Convolutional coding for bandlimited channels
http://dl-live.theiet.org/content/journals/10.1049/ip-com_19982290
The application of convolutional codes for finite intersymbol interference (ISI) channels is considered. The performance of a bandlimited communication system in the presence of additive white Gaussian noise depends on the minimum Euclidean distance <i xmlns="http://pub2web.metastore.ingenta.com/ns/">d<sub>min</sub></i> between different signal sequences. To further increase <i xmlns="http://pub2web.metastore.ingenta.com/ns/">d<sub>min</sub></i>, convolutional or trellis coding could be used. Search results indicate that the encoder with the best free distance does not always give the best coded distance when combined with an ISI channel. It has been seen from the results for several ISI channels that moderate coding gains can be obtained if the channel and the encoder are properly matched. Furthermore, it has been observed that if the channel is a maximum distance one, then the resulting coded distance is usually higher than that for a non-maximum distance channel.