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.     

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.     

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.     

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.     

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.     

Modélisation de la technique temporelle asynchrone

The evolution towards the integrated broadband communication network is undertaken by several countries. In France, system designers ofCnet in Lannion have assembled an experimental integrated broadband communication network (Ibcn)called Prelude based upon a 4.5 Gbit/s switching matrix and a new and promising transfer mode called the asynchronous time-division technique (Atd).During the project, performance evaluation studies have been carried through, aiming to validate the Atd concepts and to dimension the switching matrix. This paper reviews the Ibcn lab experiments in the world, with operational characteristics;Atd concepts are detailed and the most important features of Ibcn modelling are given. Finally, the network dimensioning is carried through giving packet size, buffers length, loss probabilities and waiting times.     

Modèle de boucles à verrouillage de phase numériques fonctionnant à faible rapport signal sur bruit

Literature is overwhelmed with many schemes of digital phase-locked loops (dpll).So a model as general as possible appeared to be badly needed. This paper reports a tentative model which allows for analysis and characterisation of a large number of dplls operating at low signal-to-noise power ratio. This model can be used for designing new circuits and evaluating their performance. The model is tested considering a first order dpll and a dpll implemented with a sequential filter. Both acquisition and tracking performances are examined.     

Propagation oblique dans les bandes de fréquences des 11 et 14 GHz

After describing the characteristics of the experimental satellite telecommunications earth-station at the Cnet,the article sets out the results of propagation measurements carried out on the one hand with the Italian experimental communications satellite Sirio I between November 1977 and November 1978, and on the other hand with the European Ots satellite between January and November 1979. 8 353 hours of useful data recording have been obtained using Sirio Iand 7 879 hours with Ots.     

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.     

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.     

Wideband and dynamic characterization of the 60GHZ indoor radio propagation — future homeWLAN architectures

This paper presents some results of a study concerning the radio propagation at 60GHZ in residential environments performed by theIETR (Electronic and Telecommunication Institute of Rennes, France). This study is part of aRNTR project named “COMMINDOR”. This exploratory project concerns the very high data rate (155 Mbit/s) and small range (about 10 m) radio communication systems for future home networks, which will permit multimedia equipment interconnection and control. The study is based on several measurement campaigns realized with a channel sounder based on the sliding correlation technique. This channel sounder has a 500MhZ bandwidth and a 2.3 ns effective time resolution. The measurements have been performed in residential furnished environments. The study of the angles-of-arrival (AOA) emphases the importance of openings (such as doors, staircase, etc.) for the radio propagation between adjacent rooms. A particular attention is paid to the influence of human activity on radio propagation. It is shown that people movements can make the propagation channel unavailable during about one second. From the characterization study of the indoor radio propagation, several recommendations concerning the deployment of the very high data rate 60GHZ wireless networks are derived.     

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.     

Circular cross-polarization measurements on earth-space paths at 11,7 GHz

Rain and icecrystal depolarization measurements were made on earthspace paths at four locations across Canada using the circularlypolarized 11.7 GHz beacon transmission from the cts satellite. Results show that when rain is the dominant depolarization mechanism, crosspolarization discrimination (xpd)is statistically related to copolar attenuation (cpa)according to xpd =U?20 log cpa with U showing improvement with increasing elevation angle. Icecrystal depolarization events, with accompanying differential phase shifts as large as 20°, were also observed. An analysis separating these events from the data demonstrates that at 11.7 GHz icecrystal depolarization degrades xpd statistics by less than 1 dB at the small percentages of time.     

A scheduler for relative delay service differentiation

We propose a new delay-based scheduler called asRD-VC (Relative Delay VirtualClock). Since it performs a delay-based service differentiation among flow aggregates, the quality at microflow level is the same as that at aggregate level. This is not easily achievable when the service differentiation is bandwidth-based or loss-based. Unlike theEDF (Earliest Deadline First) scheduler [1], our proposed scheduler self-regulates and adapts the delays according to load changes. This characteristic permits us to implement it in an AF-likePHB providing the relative quantification service in a DiffServ network. Finally, we compare our proposedrd-vc scheduler with two important existing propositions:WTP (Waiting Time Priority) [2, 3] andex-vc (Extended VirtualClock) [4]. Both these propositions are delay-based and have self-regulation property. All three schedulers (RD-VC, WTP andEX-VC) maintain the required service differentiation among aggregates and have comparable long term average performance like mean throughput per aggregate and packet loss ratio etc. However,RD-VC and WTP take an edge overEX-VC at short-term performance like jitter. Bothrd-vc andWTP have good long term and short-term performance. Our proposedrd-vc, compared to existingWTP, has two additional characteristics, i.e. unlike WTP which is limited to architectures with one queue per Qos class, it has no limitation on implementation scope (with or without separate queues per class) and it has lower complexity. This rendersRD-VC an interesting proposition.     

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.     

Protection of radiocommunications against interference from ISM equipment

Major problems of the protection of radiocommunication services against interference due to incidental radiation from industrial, scientific, and medical (ISM)equipment are considered. A theoretical model is proposed to explain the role of various factors involved. The model allows for estimation of the fraction of the transmitter coverage area lost due to unacceptable interference. Various theoretical possibilities and strategies of protection against the interference are discussed, and practical difficulties are outlined. The status of international recommendations made by the IUT (International Union of Telecommunications) and CISPR (International Special Committee on Radio Interference) in this field is presented.     

TMN implementation strategy based on OSI management standards

In order to facilitate the implementations ofTMN interface protocols/services studied inITU-T, it is very important to define profiles for supportingTMN management service. This paper proposes a concrete method for achieving this based on osi management standards as a promisingTMN implementation method. It proposes an idea of structuring theTMN ISP’S based on the structure of the osi managementISP’S. The paper discusses aTMN based on the osi managementISP’S. Finally the implementation as software is discussed and a software architecture for efficient application development is proposed.     

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.