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.     

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.     

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.     

Un environnement de conception de systèmes distribués basé sur UML

This paper introduces a new environment for developing distributed systems. It is based on theTurtle uml profile. Analysis and design phases, described in previous papers, have been extended with an additional deployment phase. In this new step,Turtle components are deployed over hardware execution nodes, and nodes are connected together throughout links,Turtle deployment diagrams are given a formal semantics inRt-lotos, therefore following the approach used forTurtle analysis and design diagrams. Moreover, the paper presents a Java code generator which outputs appropriate Java code forTurtle deployment diagrams. This code is automatically deployable on networks because it implements node communication using network protocols such asUdp orRmi. ttool, the turtle toolkit has been extended to support these new diagrams and code generators. The attack of protected data exchanged throughout securedHttp sessions serves as example.     

Multiple parallel concatenation of circular recursive systematic convolutional (Crsc) codes

This paper deals with a family of error correcting codes based on circular recursive systematic convolutional (Crsc) codes. A multiple or multidimensional parallel concatenation ofCrsc codes is proposed in order to reach minimum distances comparable to those of random codes. For information blocks of size k, minimum distances as large as k/4, for a 1/2 coding rate, may be obtained if the code dimension is raised to 4 or 5. Such codes can be decoded using an iterative (“turbo”) process relying on the extrinsic information concept.     

Décodage Syndromique à Complexité Réduite Pour Codes Convolutifs

The decoding of convolutional codes in the maximum likelihood sense is carried out in a traditional way with the Viterbi algorithm (Va). We proposed a soft and hard input decoder where theVa, associated with an relevant metric, is applied to identify the error vector rather than the information message. In this paper, we show that, with this type of decoding, the exhaustive computation of a majority ofAcs (Add Compare Select) is unnecessary. Moreover, we show that optimal performance is achieved in the case of a hard input decoder, and that performance closed to the optimum is achieved in the case of a soft input decoder, while offering of a reduction of the complexity which is all the more important than the Ec/No ratio is high (e. g. for ratio Ec/No greater than 3 dB, more than 80 % of theAcs can be avoided). We also propose an algorithm allowing rejecting a frame without having to carry out any iteration of theVa.     

Iterative low-complexity receiver for the UMTS downlink

In this paper, an iterative low-complexity receiver is proposed for Code Division Multiple Access (cdma) systems with small spreading factors. Theumts (Universal Mobile Télecommunication System) radio interface based oncdma has been designed to offer a wide range of data rates using variable spreading factors. High data rate services are obtained by using small spreading factors. For such services, the spreading sequences have bad autocor-relation properties causing the degradation of the Rake receiver performance because of the InterSymbol Interférence (isi). In order to improve the receiver performance, we propose to add a Decision Feedback Sequence Estimation (dfse) equalizer at the Rake receiver output. Thedfse is a low complexity equalizer which is able to take into accounta priori probability ratios and to deliver a posteriori probability ratios on bits in order to exchange soft information with the channel decoder, so that the proposed receiver benefits from the turbo-processing gains. Channel estimation is also treated in an iterative fashion. The complete receiver is well suited to theumts downlink system as it drastically reduces theisi while keeping a reasonable computational complexity.     

Bounds on the error probability ofml decoding for block and turbo-block codes

The performance of either structured or random turbo-block codes and binary, systematic block codes operating over the additive white Gaussian noise (Awgn) channel, is assessed by upper bounds on the error probalities of maximum likelihood (Ml) decoding. These bounds on the block and bit error probability which depend respectively on the distance spectrum and the input-output weight enumeration function (Iowef) of these codes, are compared, for a variety of cases, to simulated performance of iterative decoding and also to some reported simulated lower bounds on the performance ofMl decoders. The comparisons facilitate to assess the efficiency of iterative decoding (as compared to the optimalMl decoding rule) on one hand and the tightness of the examined upper bounds on the other. We focus here on uniformly interleaved and parallel concatenated turbo-Hamming codes, and to that end theIowefs of Hamming and turbo-Hamming codes are calculated by an efficient algorithm. The usefulness of the bounds is demonstrated for uniformly interleaved turbo-Hamming codes at rates exceeding the cut-off rate, where the results are compared to the simulated performance of iteratively decoded turbo-Hamming codes with structured and statistical interleavers. We consider also the ensemble performance of ‘repeat and accumulate’ (Ka) codes, a family of serially concatenated turbo-block codes, introduced by Divsalar, Jin and McEliece. Although, the outer and inner codes possess a very simple structure: a repetitive and a differential encoder respectively, our upper bounds indicate impressive performance at rates considerably beyond the cut-off rate. This is also evidenced in literature by computer simulations of the performance of iteratively decodedRa codes with a particular structured interleaver.     

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.     

Predictive tree encoding of still images with vector quantization

The superior performance of tree encoding and vector quantization (vq) over scalar quantization (sq) is well known. However, large constraint length (in tree encoding) and large vector length (invq), which are required for a close-to-optimum performance, are limited by the exponentially growing complexity of the source encoder. The intention of our work has been to combine tree encoding andvq, each of moderate complexity, to preserve the advantages of both block and convolutional codes. This principle was applied to the prediction error signals of images as produced by non-adaptive and adaptive linear predictors.     

Advanced decoding algorithms for Reed-Solomon/Convolutional concatenated codes

The evaluation of the union bound for theber of Reed-Solomon/Convolutional concatenated codes indicates that their performance might largely improve through the application of soft iterative decoders. This paper presents an iterative decoding algorithm for concatenated codes consisting of an outer Reed-Solomon code, a symbol interleaver and an inner convolutional code. The performance improvement for iterative and non-iterative decoders is evaluated. Existing solutions for the different decoding stages and their interfaces are discussed and their performance is compared. A new procedure is proposed to define the feedback signal from the output of the Reed-Solomon decoder to the input of the convolutional decoder, which captures the reliability information that can be inferred from errors-and-era-suresrs decoders and includes the “state pinning” approach as a particular case. The decoding schemes are applied to the specificdvb-s concatenated code.     

Design of a 30 Mbit/s, flexible frame length turbo-block-decoder

In this article we present the first prototype of hardware implementation of the new modified turbo block decoder, where the termination of all the used soft-in-soft-out decoders is possible. We also enlighten the flexibility of the decoder and the application of same hardware in systems where different block lengths are used. Simulated and measured results in simple awgn channel and in median surroundings usingOfdm are presented.     

Codage conjoint source-canal des paramètres lsf du codeur FS1016: application de la quantification vectorielle par treillis optimisée pour un canal bruité

In this paper, we present an optimized trellis coded vector quantization (tcvq) coding system designed for the effective and robust coding of lsf spectral parameters at low bit rate. The aim of this system, called at the beginning « lsf-otcvq Encoder », is to achieve a low bit rate transparent quantization of lsf parameters of the us Federal Standard (fs1016) speech coder. Once the effectiveness of our lsf-otcvq encoders (with weighted distance) was proven in the case of ideal transmissions over noiseless channel, we were interested after in the improvement of their robustnesses for real transmissions over noisy channel. To implicitly protect the transmission indices of our lsf-otcvq coders incorporated in the Fsl016, we used a joint source-channel coding carried out by the channel optimized vector quantization (covq) method. In the case of transmissions over noisy channel, we will show that our new encoding system, called “covq-lsf-otcvq Encoder”, would be able to contribute significantly in the improvement of the fs1016 performances by ensuring a good coding robustness of its lsf spectral parameters.     

A Rate \text{ M }_{\mathrm{T}} Full Diversity STF Block Coded 4\times 4 MIMO-OFDM System with Reduced Complexity

Multiple input multiple output (MIMO) communication systems with orthogonal frequency division multiplexing (OFDM) has a great role to play for 4G broadband wireless communications. In this paper, a space time frequency (STF) code is presented with reduced decoder complexity and to achieve code rate $\text{ M }_\mathrm{T}$ with full diversity of $\text{ M }_{\mathrm{T}} \text{ M }_{\mathrm{R}} \text{ N }_{\mathrm{b}}$ L i.e., product of number of transmit antennas ( $\text{ M }_\mathrm{T}$ ), receive antennas $(\text{ M }_{\mathrm{R}})$ , fading blocks $(\text{ N }_{\mathrm{b}})$ and channel taps (L). The maximum achievable diversity with high rate of STF block coded MIMO-OFDM is analyzed and verified by simulation results. The decoder complexity is resolved by employing several approaches like maximum likelihood (ML), sphere decoder (SD) and array processing. The performance of STF code is compared with existing layered algebraic STF code in terms of decoder complexity and bit error rate (BER). Further, the closed form expressions for BER performance of STFBC MIMO-OFDM systems are derived and evaluated for frequency selective block fading channels with MPSK constellations.     

Turbo soft combining hybrid ARQ techniques: Theory and application to 3G wireless networks

This paper deals with the application of Hybrid Automatic Repeat reQuest (h-arq) techniques to allow reliable data communications in wireless 3G networks. Basically, retransmission of coded data is endowed with soft combining schemes applied, respectively, to packet replicas, or to decoding algorithm outputs. In particular, the proposed coding scheme takes advantage of error correction capabilities of the turbo codes, while the combining algorithm follows the diversity approach. The performance of the proposedh-arq schemes has been derived by means of a suitable analytical approach and numerical simulations in the case of a typical umts environment. The results highlight the good behavior of the proposed scheme in term of error rate, throughput, packet delivery delay and power reduction.     

Bandwidth sharing under the assured forwarding Per-Hop behavior

The DiffServ’s Assured Forwarding (af) Per-Hop Behavior (phb) Group defines a differentiated forwarding of packets in four independent classes, each class having three levels of drop precedence. Specific end-to-end services based on thisphb are still being defined. A particular type of service that could assure a given rate to a traffic aggregate has been outlined elsewhere. In such a service, a fair distribution of bandwidth is one of the main concerns. This paper presents experimental work carried out to evaluate howaf distributes bandwidth among flows under different load conditions and traffic patterns. We focused on the effect that marking mechanisms have on bandwidth sharing among flows within a singleaf class. The traffic types we used includeudp flows, individual and aggregatedtcp flows, mix oftcp andudp, tcp sessions with heterogeneous round-trip times, as well as color-blind and color-aware re-marking at the aggregation point fortcp flows. Tests were performed on real and simulated networks. We have found certain conditions under whichaf distributes bandwidth fairly among nonadaptiveudp flows andtcp aggregates. Finally, we evaluate a basic rule for setting the parameters of the two-rate Three-Color Marker conditioning algorithm (trtcm) in order to achieve a better bandwidth distribution fortcp flows.     

Architecture d’un réseau actif à base de services Web

This paper presents a novel active architecture for building and deploying network services:aswa, Web Services based Active network Architecture. At the architectural level,aswa defines an active node whose functionalities are divided into the Node Operating System, the Execution Environment, and the Active Applications. At the implementation level,aswa is a Web Services based platform where new components could be added and deployed, in order to dynamically modify network nodes behavior. Applications can be developed with any language and communicate across heterogeneous environments, and across Internet and Intranet structures. At the deployment levelaswa uses an active node approach, and offers a controlled deployment mode. In terms of security, Authentication of deployed code and protection of the nodes is achieved by the use ofhttps and the header extensions of thesoap envelope. Finally to validate this architecture,aswa defines a Firewall as an Active Application to secure the code deployment.     

Robust time-frequency distributions based on the robust short time fourier transform

Design of time-frequency distributions (Tfds) that are robust to the impulse noise influence is considered. The robustTfds based on the robust short-time Fourier transform (Stft) are proposed. An efficient procedure to evaluate the robustStft is given. RobustTfds based on the robustStft have better energy concentration around the signal instantaneous frequency (If) than the robustStft itself. Also, theseTfds are more resistant to higher impulse noise than the robustTfds obtained using the local autocorrelation function (Laf) based minimization problem.     

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.     

Characterization and evaluation of TCP and UDP-based transport on real networks

StandardTcp (RenoTcp) does not perform well on fast long distance networks, due to its AMD congestion control algorithm. In this paper we consider the effectiveness of various alternatives, in particular with respect to their applicability to a production environment. We then characterize and evaluate the achievable throughput, stability and intra-protocol fairness of differentTcp stacks (Scalable,Hstcp,Htcp, FastTcp, Reno,Bictcp, hstcp-lp andLtcp) and aUdp based application level transport protocol (Udtv2) on both production and testbed networks. The characterization is made with respect to both the transient traffic (entry and exit of different streams) and the steady state traffic on production Academic and Research networks, using paths withRtts differing by a factor of 10. We also report on measurements made with 10 Gbit/secNics with and withoutTcp Offload Engines, on 10 Gbit/s dedicated paths set up forSc2004.