Improving parallel EKF-based nonlinear channel equalization using unscented transformation

The paper presents a new review of parallel Kalman filtering for nonlinear channel equalization. A Network of Extended Kalman Filters (nekf) has already been suggested for this purpose. This equalizer gives recursively a minimum mean squared error (mmse) estimation of a sequence of transmitted symbols according to a state formulation of a digital communication scheme. It is essentially based on two mechanisms: the approximation of the non Gaussiana posteriori probability density function (pdf) of the symbol sequence by a Weighted Gaussian Sum (wgs); and the local linearization of the nonlinear channel function for each branch of the network. Since the linearization, bearing on scattered symbol states, is one of the major limitations of thenekf, a new Kalman filtering approach, the Unscented Kalman Filter (ukf) suggested by Julier and Uhlman is considered in this paper for an interesting adaptation to the equalization context. Theukf algorithm is based on the equations of a Kalman filter, as the optimal linear minimum variance estimator, and on determining conditional expectations based on a kind of deterministic Monte-Carlo simulations. The new equalizer referred to as the Network ofukf (nukf), thus combines density approximation by awgs and the Unscented Transformation (ut) principle to circumvent the linearization brought within eachekf and is shown to perform better than thenekf based equalizer for severe nonlinear channels. Also, an adaptive version of thenukf is developed using the k-means clustering algorithm for noise-free channel output identification, since thenukf-based algorithm does not require the knowledge of the channel nonlinearity model.     

Simple iterative receivers for MIMO LP-OFDM systems

In this paper,Lp-ofdm is combined with differentMimo schemes in order to improve performance in terms of diversity gain and to exploit capacity brought by theMimo channel. The original contribution is the development of a generic iterative receiver designed forLp mimo transmission able to work whatever the antenna configuration and the spatial coding scheme. By using a globalMmse criterion, interference terms coming from space-time coding and linear precoding are jointly treated leading to a very good trade-off between performance and complexity compared to trellis based detectors particularly for high order modulations, high number of antennas and/or large size of precoding matrices.     

Parallel interference cancellation efficiency for noisy signals in O-CDMA system

The aim of this paper is to evaluate the robustness of Parallel Interference Cancellation (Pic) to noise contribution for an optical Code Division Multiple Access system. The theoretical expression of thePic error probability is developed in the case of white additive Gaussian noise. From theoretical analysis, we show that, even with noise contribution, thePic receiver outperforms the Conventional Correlation Receiver (Ccr). Moreover, the results highlight that, for a given performance, thePic receiver relaxes not only the constraint on the code length, but also the Signal to Noise Ratio compared toCcr. Particularly, this is proofed in access network context, i.e. 30 users withBer lt; 10^?9.     

Iterative successive interference cancellation for multi-user DS/CDMA detectors in multipath channels

This paper deals with uplink Direct-Sequence Code Division Multiple Access (DS-CDMA) transmissions over mobile radio channels. A new interference cancellation scheme for multiuser detection, calledSIC/RAKE, is presented. It is based on a modified multistage Successive Interference Cancellation (sic) structure that enables efficient detection in multipath propagation environments, thanks to a single userRAKE receiver incorporated in each unit of thesic structure. Furthermore, a modified version of thesic structure, calledSIC/MMSE, that ensures convergence to theMMSE detector rather than to the decorrelating detector has been suggested. The convergence of theSIC/RAKE andSIC/MMSE methods is proved. Simulation results for the Universal Mobile Telecommunication System (UMTS) have been carried out for flat fading Rayleigh multipath channels, showing that the proposed detector is resistant to the near-far effect and that low performance loss is obtained compared to the single-user bound.     

Securing communications overAtm networks: the remoteAtm private networks interconnection example

When remoteAtm sites communicate through anAtm public network, a number of security problems arise, such as hacking, eavesdropping and traffic tampering. This paper proposes three contributions to these security problems. Firstly, risks due toAtm technology usage are detailed. Secondly, a survey of existing techniques aiming at securingAtm communications is presented with emphasis on theAtm Forum’s security specifications. Thirdly, a new solution called Safe (which stands for Solution for anAtm Frequent communications Environment) developed in the Démostène project is described. Safe realizes both firewall’s filtering functions and communications protection over theAtm network. The main idea of Safe is to use signaling (Uni 3.1) as a means to exchange security information over the network. This idea has been implemented and introduced to theAtm Forum.     

Modulation par codage spectral

Using the time-frequency duality principle, the author gives a relation between the so-called Pulse Code Modulation system (Pcm)and a system in which the spectral lines are coded in a similar way (Fcm or Frequency Code Modulation system). The latter appears to be quite comparable to a four phase digital modem, the carrier frequencies of which are multiples of a common basic frequency. It is showed that a Fcm frequency coded multiplexing device may be built, which is quite comparable to a Pcm equipment, as far as the bandwith requirement is concerned. The Fcm terminal equipment is much more expensive and sophisticated than the Pcm one, but the Fcm line amplifiers are simpler than the Pcm regenerative repeaters. The development of the Fcm system relies upon further technical and economical progress of the technology of digital discrete Fourier transformer operating at very high speed.     

Measurement-based admission control in UMTS

In this paper, we develop an efficient Call Admission Control (cac) algorithm forumts systems. We first introduce the expressions that we developed for Signal-to-Interference (sir) for both uplink and downlink, to obtain a novelcac algorithm that takes into account, in addition tosir constraints, the effects of mobility, coverage as well as the wired capacity behind the base station, for the uplink, and the maximal transmission power of the base station, for the downlink. As of its implementation, we investigate the measurement-based approach as a means to predict future, both handoff and new, call arrivals and thus manage different priority levels depending on a tunable coefficient. Compared to classicalcac algorithms, ourcac mechanism achieves better performance in terms of outage probability and QoS management.     

Authentication in wireless networks: state of the art and integration with smart card

Wirelesslan have spread very quickly over the past few years. Demand for wireless access tolans has raised due to new mobile computing devices, such as laptops and personal digital assistants, and a desire for seamless and permanent connections to networks. Nevertheless a lot of security issues remain and stop its deployment in corporations. One of the most important issues is the authentication of a terminal to an Access Point. We propose an interface to integrate the Extensible Authentication Protocol into smart cards and will show that smart cards could constitute the de-facto device for authentication in Wirelesslan as they are forgsm and will be forumts (Universal Mobile Telecommunication System).     

Optimisation de la quantification vectorielle codée par treillis: application au codage des paramètres LSF

Speech coders operating at low bit rates necessitate efficient encoding of the linear predictive coding (Lpc) coefficients. Line spectral Frequencies (Lsf) parameters are currently one of the most efficient choices of transmission parameters for theLpc coefficients. In this paper, an optimized trellis coded vector quantization (Tcvq) scheme for encoding theLsf parameters is presented. When the selection of a proper distortion measure is the most important issue in the design and operation of the encoder, an appropriate weighted distance measure has been used during theTcvq construction process. We further applied the optimizedTcvq system for encoding theLsf parameters of the us Federal Standard (Fs1016) 4.8 kbps speech coder. At lower bit rates, objective and subjective evaluation results show that the incorporatedLsf tcvq encoder performs better than the 34 bits/frameLsf scalar quantizer used originally in the fs1016 coder. The subjective tests reveal also that the 27 bit/frame scheme produces equivalent perceptual quality to that when theLsf parameters are unquantized.     

A MC-CDMA system analysis in a software radio context

This paper presents a Multi-Carrier Code Division Multiple Access (Mc-Cdma) system analysis in a software radio context. Based on a combination of multi-carrier modulation and code division multiple access,Mc-Cdma benefits from the main advantages from both schemes: high spectral efficiency, high flexibility, multiple access capabilities, etc. It is firstly shown why, nowadays,Mc-Cdma is undoubtedly a high potential candidate for the air interface of the 4G cellular networks. TheMc-Cdma concept and the block-diagrams of the transmitter and the receiver are presented first. Afterwards, the technical issues concerning the processing devices for the implementation ofMc-Cdma systems in a software radio context are analysed. The advantages and disadvantages of Digital Signal Processors (Dsps) and Field Programmable Gate Arrays (Fgpas) components are discussed. The implementation ofMc-Cdma systems and the integration of signal processing algorithms as Fast Hadamard Transform (Fht) and Inverse Fast Fourier Transform (Ifft) are considered and analysed for the first time. Finally, implementation results with a mixed prototyping board are presented. Then, it is shown that a new combination of the flow graphs ofFht andIfft leads to interesting computation savings and that hardware structures asFgpas are more adapted thanDsps to those intensive computation functions. Finally, for the completeMc-Cdma modem implementation, the necessity of a Co-Design methodology is highlighted in order to obtain the best matching between algorithms and architecture.     

LDPC-based space-time coded OFDM systems: Turbo-EM receiver design with channel state information guessing algorithms

In this paper we study some turbo receiver architectures employing low-density parity check (Ldpc) codes together with orthogonal frequency division multiplexing (Ofdm) for high data rate wireless transmissions. Different demodulation schemes based on expectation-maximization (Em) algorithm are studied along with the channel impulse response (Em) algorithms. We studied differentCir guessing algorithms including the EM-based algorithms such as a space-alternating generalized expectation-maximization algorithm (Sage). It is shown that the proposed turbo-Em receiver employing a soft maximum a posteriori (Map)Em demodulator and a belief propagationLdpc decoder can perform within 1 dB from the ergodic capacity of the studiedMimo ofdm channels. Besides, we find that a suboptimum structure based on a soft interference cancellationMmse filtering demodulator exhibits negligible loss in non-correlated fadingMimo channels but suffer extra performance loss in highly correlatedMimo channels.     

IP-based transport of voice traffic in the UMTS radio access network: analytical study and empirical validation

In this paper, we investigate theIp protocol as a transport option for the user traffic in the UMTS Terrestrial Radio Access Network (Utran), where stringent delay bounds are to be met for both real-time and non real-time traffic. We focus on real-time voice traffic and present an analytical model for the multiplexing and transport of voice channels in theUtran usingIp. The novelty of our model is that it analytically includes and quantifies the performance of the timer used in multiplexing arriving Frame Protocol (Fp) frames into largerIp packets. We then validate our work through empirical results on a test-bed emulating theUtran transport functionalities. We show the trade-offs between performance, in terms of delay and link utilization, and quantify optimal values for the timer as well as the number ofFp frames perIp packet for a given output link capacity.     

Technical and economic aspects of the different fibre fabrication processes

The processes for making preforms for hard glass fibres are the Vad process, the Ovd process, the Mcvd (Pmcvd) process and the Pcvd process. This paper discusses the processes themselves, the optical properties of multimode, single-mode and high Na fibres made with these processes and the potentials of the different processes. For really large scale applications it will be necessary to reduce the costs of manufacturing these fibres. The factors that determine these costs are discussed and ways in which they can be reduced are indicated.     

On the stochastic timed Pétri nets model and its application to the DQDB protocol

The goal of this paper is twofold : at first, to present the stochastic timed Pétri nets model by emphasizing its ability to express the main characteristics of real time distributed systems (parallelism, synchronization by message exchange, time attributes and in particular time constraints) and to allow both qualitative and quantitative analysis (concept of randomized states graph); second, to show a modeling methodology based on this model that allows a rigorous modeling of theDqdb protocol (Qa access, bwb mechanism role,Pa access influence). A new contribution to theQa mechanism has been done (1).     

On turbo encodedBicm

The superior performance of the binary turbo codes has stimulated vigorous efforts in generating bandwidth efficient modulation schemes adhering to these codes. Several approaches for the integration of turbo-coding and modulation have emerged in recent years but none seem to dominate. In the bit interleaved coded modulation (Bicm) scheme is used to achieve high bandwidth and power efficiency, while separating coding and modulation. As is now well known, theBicm scheme achieves capacity remarkably close to the constellation channel capacity. The turbo-Bicm scheme enjoys high coding diversity (well suited for fading channels), high flexibility as well as design and implementation simplicity, while maintaining good power efficiency. The system comprises one standard turbo code, an interleaver, a mapper and a modulator at the transmitter, corresponding to a demodulator, a de-interleaver and a turbo decoder at the receiver. A modified system, which improves on performance by incorporating the demodulation in the iterative decoding procedure, is investigated, and some performance gain is demonstrated, especially for low rate codes. Information theoretic arguments for the somewhat minor potential improvement in performance are detailed. The preferred mapper and interleaver for this system are considered. Extending previous works, for higher level modulations, we analyze a system including a convolutional code, an interleaver, a differential encoder (De), a mapper and a modulator at the transmitter. As for theBpsk modulation, the serial concatenation of a convolutional code withDe outperforms the single convolutional code. The serial concatenation withDe approach is analyzed also for a turbo code, where it is found to fail in achieving performance improvement. Several structures for the serial concatenation withDe are examined. These results are substantiated through the ‘spectral thinning’ phenomena of the weight distribution of the convolutional and turbocodes.     

Analog to digital conversion: technical aspects

An ideal radio communication receiver places the analog to digital conversion just after the antenna. It is an objective in a “software radio” perspective. The available silicon technologies do not provide the performance required by this application. We are able to evaluate the present limits and the gap between these limits and the ideal solution proposed. In this paper, we describe the present possibilities in terms of receiver architectures and we deduce theAdc specifications. Then we analyse differentAdc architectures adapted to this application. The choice is mainly between pipeline and sigma- deltaAdc. We compare them in terms of power consumption and we introduce a factor of merit. The future technologies will have an impact onAdc performance. Superconductor technology applied toAdc may be a solution and it is analysed at the end of this paper.     

AIMD-based online MPLS traffic engineering for TCP flows via distributed multi-path routing

With this paper, we propose a distributed online traffic engineering architecture formpls networks. In this architecture, a primary and secondarympls lsp are established from an ingresslsr to every other egresslsr. We propose to split thetcp traffic between the primary and secondary paths using a distributed mechanism based onecn marking andaimd-based rate control. Inspired by the random early detection mechanism for active queue management, we propose a random early reroute scheme to adaptively control the delay difference between the primary and secondarylsps. Considering the adverse effect of packet reordering ontcp performance for packet-based load balancing schemes, we propose that thetcp splitting mechanism operates on a per-flow basis. Using flow-based models developed for Internet traffic and simulations, we show that flow-based distributed multi-path traffic engineering outperforms on a consistent basis the case of a single path in terms of per-flow goodputs. Due to the elimination of out-of-order packet arrivals, flow-based splitting also enhancestcp performance with respect to packet-based splitting especially for longtcp flows that are hit hard by packet reordering. We also compare and contrast two queuing architectures for differential treatment of data packets routed over primary and secondarylsps in thempls data plane, namely first-in-first-out and strict priority queuing. We show through simulations that strict priority queuing is more effective and relatively more robust with respect to the changes in the traffic demand matrix than first-in-first-out queuing in the context of distributed multi-path routing.     

Démarche de conception d’une plate-forme monopuce de réception de radiodiffusion numérique DRM

The newdrm standard for digital radio broadcast in theam band requires integrated devices for radio receivers at low cost and very low power. A chipset has been developed based on anarm9 dual-core architecture. This paper introduces the application itself, and then presents the development process followed to design such a chip covering the hardware/software partitioning, the architecture definition and all the validation issues. Therf part that partially drives the baseband is also briefly presented. The legal aspects are also investigated. Then as a conclusion, this paper presents the next steps of this development and the methodological gap in the system on chip design process.     

Mesure du Débit d’Absorption Spécifique (das) des téléphones mobiles — contribution du projetcomobio à la normalisation internationale

Human exposure to electromagnetic fields emitted by mobile phones is characterised by the Specific Absorption Rate (sar). This value corresponds to therf power absorbed by unitary mass of the equivalent tissue. It represents the basic restriction used to define limit of human exposure torf electromagnetic fields. Academic and industrial labs have contributed through thecomobio project to define international standards for mobile phones certification.     

EnhancingIEEE 802.11 performance in congested environments

IEEE 802.11 is the most deployed wireless local area networking standard nowadays. It uses carrier sense multiple access with collision avoldance (CSMA/CA) to, resolve contention between nodes. Contention windows (CW) change dynamically to adapt to the contention level: Upon each collision a node doubles itsCW to, reduce further collision risks. Upon a successful transmission theCW is reset, assuming the contention level dropped. However, contention level is more likely to change slowly, and resetting theCW causes new collisions and retransmissions before reaching the optimal value again. This wastes bandwidth and increases delays. In this paper we propose simple slowCW decrease functions and compare their performances to the legacy standard. We analyze them through simulation and show their considerable enhancement at all congestion levels and transient phases.