Subspace-based channel estimation for DS/CDMA systems exploiting pulse- shaping information.

Thesis (M.Sc.Eng.)-University of Natal, Durban, 2003.Third generation wireless systems have adopted Direct-Sequence/Code-Division Multiple-Access (DS/CDMA) as the multiple access scheme of communication. This system would typically operate in a multipath fading channel. This dissertation only deals with the task of channel estimation at the base station where the multipath delays and attenuations for each user are estimated. This information is used to aid the recovery of data that was transmitted by each user. Subspace-based algorithms are popularly used to perform the task of channel estimation because they have the desirable property of perfectly estimating the channel in a noise-free environment. In this dissertation a new subspace-based channel estimation algorithm for DS/CDMA systems is presented. The proposed algorithm is based on the Parametric Subspace algorithm by Perros-Meilhac et al. for single-user systems. The main focus of this dissertation is to convert the Parametric Subspace algorithm from a single-user system to a multi-user DS/CDMA system. It has been shown in the literature that by using information of the pulse-shaping filter in the Channel Subspace algorithm, the variance of the channel estimates is decreased. However, this has only been applied to a single-user system. There are several subspace algorithms that have been proposed for DS/CDMA systems. Most of these algorithms sample the received signal at the chip rate, making it impossible to exploit knowledge of the pulse-shaping filter in the channel estimation algorithm. In this dissertation a new subspace-based channel estimation algorithm is derived for a DS/CDMA system with multiple receive antennas, where the output is oversampled with respect to the chip rate. By oversampling the received signal, knowledge of the pulse-shaping filter is used in the channel estimation algorithm. It is shown that the variance of the channel estimate for the proposed subspace algorithm is less than the Torlak/Xu subspace algorithm that does not exploit information of the pulse-shaping filter. A mathematical expression of the mean square error of estimation for the new algorithm is also derived. It was shown that the analytic expression provides a good approximation of the actual MSE for high SNR. The Parametric Subspace Delay Estimation (PSDE) algorithm was developed by Perros-Meilhac et al. to estimate the multipath delays introduced by the communications channel. The limitation of the PSDE algorithm is that the performance of the algorithm deteriorates as the power of the multipath signals decrease with increasing delay time. This dissertation proposes a modified version of the PSDE algorithm, called the Modified Parametric Subspace Delay Estimation (MPSDE) algorithm, which performs better than the PSDE algorithm in an environment where the power of the multipath signals varies. The final part of this dissertation discusses the Torlak/Xu channel estimation algorithm and the Bensley/Aazbang delay estimation algorithm. In order to compare the performance of these two subspace algorithms, the Torlak/Xu algorithm is converted to a delay estimation algorithm that is called the Parametric TX algorithm. The performance of the Bensley/Aazbang delay estimation algorithm and the proposed Parametric TX algorithm are compared and it is shown that the Parametric TX algorithm offers the better performance

MMSE equalizers and precoders in turbo equalization.

Thesis (M.Sc.Eng.)-University of Natal, Durban, 2003.Transmission of digital information through a wireless channel with resolvable multipaths or a bandwidth limited channel results in intersymbol interference (1SI) among a number of adjacent symbols. The design of an equalizer is thus important to combat the ISI problem for these types of channels and hence provides reliable communication. Channel coding is used to provide reliable data transmission by adding controlled redundancy to the data. Turbo equalization (TE) is the joint design of channel coding and equalization to approach the achievable uniform input information rate of an ISI channel. The main focus of this dissertation is to investigate the different TE techniques used for a static frequency selective additive white Gaussian noise (AWGN) channel. The extrinsic information transfer (EXIT) chart is used to analyse the iterative equalization/decoding process and to determine the minimum signal to noise ratio (SNR) in order to achieve convergence. The use of the Minimum Mean Square Error (MMSE) Linear Equalizer (LE) using a priori information has been shown to achieve the same performance compared with the optimal trellis based Maximum A Posterior (MAP) equalizer for long block lengths. Motivated by improving the performance of the MMSE LE, two equalization schemes are initially proposed: the MMSE Linear Equalizer with Extrinsic information Feedback (LE-EF (1) and (U)). A general structure for the MMSE LE, MMSE Decision Feedback Equalizer (DFE) and two MMSE LE-EF receivers, using a priori information is also presented. The EXIT chart is used to analyse the two proposed equalizers and their characteristics are compared to the existing MAP equalizer, MMSE LE and MMSE DFE. It is shown that the proposed MMSE LE-EF (1) does have an improved performance compared with the existing MMSE LE and approaches the MMSE Linear Equalizer with Perfect Extrinsic information Feedback (LE-PEF) only after a large number of iterations. For this reason the MMSE LE-EF is shown to suffer from the error propagation problem during the early iterations. A novel way to reduce the error propagation problem is proposed to further improve the performance of the MMSE LE-EF (I). The MAP equalizer was shown to offer a much improved performance over the MMSE equalizers, especially during the initial iterations. Motivated by using the good quality of the MAP equalizer during the early iterations and the hybrid MAP/MMSE LE-EF (l) is proposed in order to suppress the error propagation problem inherent in the MMSE LE-EF (I). The EXIT chart analysis reveals that the hybrid MAP/MMSE LE-EF (l) requires fewer iterations in order to achieve convergence relative to the MMSE LE-EF (l). Simulation results demonstrate that the hybrid MAP/MMSE LE-EF (I) has a superior performance compared to the MMSE LE-EF (I) as well as approaches the performance of both the MAP equalizer and MMSE LE-PEF at high SNRs, at the cost of increased complexity relative to the MMSE LEEF (I) receiver. The final part of this dissertation considers the use of precoders in a TE system. It was shown in the literature that a precoder drastically improves the system performance. Motivated by this, the EXIT chart is used to analyse the characteristics of four different precoders for long block lengths. It was shown that using a precoder results in a loss in mutual information during the initial equalization stage. However" we show by analysis and simulations that this phenomenon is not observed in the equalization of all precoded channels. The slope of the transfer function, relating to the MAP equalization of a precoded ISI channel (MEP), during the high input mutual information values is shown to play an important role in determining the convergence of precoded TE systems. Simulation results are presented to show how the precoders' weight affects the convergence of TE systems. The design of the hybrid MAP/MEP equalizer is also proposed. We also show that the EXIT chart can be used to compute the trellis code capacity of a precoded ISI channel