This thesis investigates channel-coded physical layer network coding, in which the relay directly transforms the noisy superimposed channel-coded packets received from the two end nodes, to the network-coded combination of the source packets. This is in contrast to the traditional multiple-access problem, in which the goal is to obtain each message explicitly at the relay. Here, the end nodes $A$ and $B$ choose their symbols, $S_A$ and $S_B$, from a small non-binary field, $\mathbb{F}$, and use non-binary PSK constellation mapper during the transmission phase. The relay then directly decodes the network-coded combination ${aS_A+bS_B}$ over $\mathbb{F}$ from the noisy superimposed channel-coded packets received from two end nodes. Trying to obtain $S_A$ and $S_B$ explicitly at the relay is overly ambitious when the relay only needs $aS_B+bS_B$. For the binary case, the only possible network-coded combination, ${S_A+S_B}$ over the binary field, does not offer the best performance in several channel conditions. The advantage of working over non-binary fields is that it offers the opportunity to decode according to multiple decoding coefficients $(a,b)$. As only one of the network-coded combinations needs to be successfully decoded, a key advantage is then a reduction in error probability by attempting to decode against all choices of decoding coefficients. In this thesis, we compare different constellation mappers and prove that not all of them have distinct performance in terms of frame error rate. Moreover, we derive a lower bound on the frame error rate performance of decoding the network-coded combinations at the relay. Simulation results show that if we adopt concatenated Reed-Solomon and convolutional coding or low density parity check codes at the two end nodes, our non-binary constellations can outperform the binary case significantly in the sense of minimizing the frame error rate and, in particular, the ternary constellation has the best frame error rate performance among all considered cases.

In wireless communications, knowledge of the channel coefficients is required for coherent demodulation. In this thesis, a blind channel estimation method based on the Lloyd-Max algorithm is proposed for single-tap fading channels. The algorithm estimates the constellation points for the received signal using an iterative least squares approach. The algorithm is investigated for fast-frequency hopping systems with small block lengths and operating under partial-band and partial-time jamming for both detecting the jammer and estimating the channel. The performance of the Lloyd-Max channel estimation algorithm is compared to the performance of pilot-based channel estimation algorithms which also use the least squares approach and non-coherent demodulation and decoding.

This thesis investigates channel-coded physical layer network coding, in which the relay directly transforms the noisy superimposed channel-coded packets received from the two end nodes, to the network-coded combination of the source packets. This is in contrast to the traditional multiple-access problem, in which the goal is to obtain each message explicitly at the relay. Here, the end nodes $A$ and $B$ choose their symbols, $S_A$ and $S_B$, from a small non-binary field, $\mathbb{F}$, and use non-binary PSK constellation mapper during the transmission phase. The relay then directly decodes the network-coded combination ${aS_A+bS_B}$ over $\mathbb{F}$ from the noisy superimposed channel-coded packets received from two end nodes. Trying to obtain $S_A$ and $S_B$ explicitly at the relay is overly ambitious when the relay only needs $aS_B+bS_B$. For the binary case, the only possible network-coded combination, ${S_A+S_B}$ over the binary field, does not offer the best performance in several channel conditions. The advantage of working over non-binary fields is that it offers the opportunity to decode according to multiple decoding coefficients $(a,b)$. As only one of the network-coded combinations needs to be successfully decoded, a key advantage is then a reduction in error probability by attempting to decode against all choices of decoding coefficients. In this thesis, we compare different constellation mappers and prove that not all of them have distinct performance in terms of frame error rate. Moreover, we derive a lower bound on the frame error rate performance of decoding the network-coded combinations at the relay. Simulation results show that if we adopt concatenated Reed-Solomon and convolutional coding or low density parity check codes at the two end nodes, our non-binary constellations can outperform the binary case significantly in the sense of minimizing the frame error rate and, in particular, the ternary constellation has the best frame error rate performance among all considered cases.

The main subject of this thesis is creating a programme, used for channel coding demonstration. This programme will be used for teaching purposes. The programme contains various codes from simple ones, to those which almost reach Shanon’s channel capacity theorem. Specifically these are the Hamming code, cyclic code, convolutional code and LDPC code. These functions are based on theoretical background described in this thesis and have been programed in Matlab. Practical output of this thesis is user interface, where the user is able to input information word, simulate transmission through the transmission channel and observe coding and decoding for each code. This thesis also contains a comparison between individual codes, concerning bit-error rate depending on SNR and various parameters. There is a computer lab with theoretical background, assignment and sheets for convenient accomplishment of each task.

Optimising the throughput performance for wireless networks is one of the challenging tasks in the objectives of communication engineering, since wireless channels are prone to errors due to path losses, random noise, and fading phenomena. The transmission errors will be worse in a multihop scenario due to its accumulative effects. Network Coding (NC) is an elegant technique to improve the throughput performance of a communication network. There is the fact that the bit error rates over one modulation symbol of 16- and higher order- Quadrature Amplitude Modulation (QAM) scheme follow a certain pattern. The Scattered Random Network Coding (SRNC) system was proposed in the literature to exploit the error pattern of 16-QAM by using bit-scattering to improve the throughput of multihop network to which is being applied the Random Linear Network Coding (RLNC). This thesis aims to improve further the SRNC system by using Forward Error Correction (FEC) code; the proposed system is called Joint RLNC and FEC with interleaving. The first proposed system (System-I) uses Convolutional Code (CC) FEC. The performances analysis of System-I with various CC rates of 1/2, 1/3, 1/4, 1/6, and 1/8 was carried out using the developed simulation tools in MATLAB and compared to two benchmark systems: SRNC system (System-II) and RLNC system (System- III). The second proposed system (System-IV) uses Reed-Solomon (RS) FEC code. Performance evaluation of System IV was carried out and compared to three systems; System-I with 1/2 CC rate, System-II, and System-III. All simulations were carried out over three possible channel environments: 1) AWGN channel, 2) a Rayleigh fading channel, and 3) a Rician fading channel, where both fading channels are in series with the AWGN channel. The simulation results show that the proposed system improves the SRNC system. How much improvement gain can be achieved depends on the FEC type used and the channel environment.

Optimising the throughput performance for wireless networks is one of the challenging tasks in the objectives of communication engineering, since wireless channels are prone to errors due to path losses, random noise, and fading phenomena. The transmission errors will be worse in a multihop scenario due to its accumulative effects. Network Coding (NC) is an elegant technique to improve the throughput performance of a communication network. There is the fact that the bit error rates over one modulation symbol of 16- and higher order- Quadrature Amplitude Modulation (QAM) scheme follow a certain pattern. The Scattered Random Network Coding (SRNC) system was proposed in the literature to exploit the error pattern of 16-QAM by using bit-scattering to improve the throughput of multihop network to which is being applied the Random Linear Network Coding (RLNC). This thesis aims to improve further the SRNC system by using Forward Error Correction (FEC) code; the proposed system is called Joint RLNC and FEC with interleaving. The first proposed system (System-I) uses Convolutional Code (CC) FEC. The performances analysis of System-I with various CC rates of 1/2, 1/3, 1/4, 1/6, and 1/8 was carried out using the developed simulation tools in MATLAB and compared to two benchmark systems: SRNC system (System-II) and RLNC system (System- III). The second proposed system (System-IV) uses Reed-Solomon (RS) FEC code. Performance evaluation of System IV was carried out and compared to three systems; System-I with 1/2 CC rate, System-II, and System-III. All simulations were carried out over three possible channel environments: 1) AWGN channel, 2) a Rayleigh fading channel, and 3) a Rician fading channel, where both fading channels are in series with the AWGN channel. The simulation results show that the proposed system improves the SRNC system. How much improvement gain can be achieved depends on the FEC type used and the channel environment. / Indonesian Government and the University of Bradford

During the race towards the4th generation (4G) cellular-based digital communication systems, a growth in the demand for high capacity, multi-media capable, improved Quality-of-Service (QoS) mobile communication systems have caused the developing mobile communications world to turn towards betterMultiple Access (MA) techniques, like Code Division Multiple Access (CDMA) [5]. The demand for higher throughput and better QoS in future 4G systems have also given rise to a scheme that is becoming ever more popular for use in these so-called ‘bandwidth-on-demand’ systems. This scheme is known as adaptive channel coding, and gives a system the ability to firstly sense changes in conditions, and secondly, to adapt to these changes, exploiting the fact that under good channel conditions, a very simple or even no channel coding scheme can be used for Forward Error Correction(FEC). This will ultimately result in better system throughput utilization. One such scheme, known as incremental redundancy, is already implemented in the Enhanced Data Rates for GSM Evolution (EDGE) standard. This study presents an extensive simulation study of a Multi-User (MU), adaptive channel coded Direct Sequence Spread Spectrum Multiple Access (DS/SSMA) communication system. This study firstly presents and utilizes a complex Base Band(BB) DS/SSMA transmitter model, aimed at user data diversity [6] in order to realize the MU input data to the system. This transmitter employs sophisticated double-sideband (DSB)Constant-Envelope Linearly Interpolated Root-of-Unity (CE-LI-RU) filtered General Chirp-Like (GCL) sequences [34, 37, 38] to band limit and spread user data. It then utilizes a fully user-definable, complex Multipath Fading Channel Simulator(MFCS), first presented by Staphorst [3], which is capable of reproducing all of the physical attributes of realistic mobile fading channels. Next, this study presents a matching DS/SSMA receiver structure that aims to optimally recover user data from the channel, ensuring the achievement of data diversity. In order to provide the basic channel coding functionality needed by the system of this study, three simple, but well-known channel coding schemes are investigated and employed. These are: binary Hamming (7,4,3) block code, (15,7,5) binary Bose-Chadhuri-Hocquenghem (BCH) block code and a rate 1/3 <i.Non-Systematic (NS) binary convolutional code [6]. The first step towards the realization of any adaptive channel coded system is the ability to measure channel conditions as fast as possible, without the loss of accuracy or inclusion of known data. In 1965, Gooding presented a paper in which he described a technique that measures communication conditions at the receiving end of a system through a device called a Performance Monitoring Unit (PMU) [12, 13]. This device accelerates the system’sBit Error Rate (BER) to a so-called Pseudo Error Rate(PER) through a process known as threshold modification. It then uses a simple PER extrapolation algorithm to estimate the system’s true BER with moderate accuracy and without the need for known data. This study extends the work of Gooding by applying his technique to the DS/SSMA system that utilizes a generic Soft-Output Viterbi Algorithm(SOVA) decoder [39] structure for the trellis decoding of the binary linear block codes [3, 41-50], as well as binary convolutional codes mentioned, over realistic MU frequency selective channel conditions. This application will grant the system the ability to sense changes in communication conditions through real-time BER measurement and, ultimately, to adapt to these changes by switching to different channel codes. Because no previous literature exists on this application, this work is considered novel. Extensive simulation results also investigate the linearity of the PER vs. modified threshold relationship for uncoded, as well as all coded cases. These simulations are all done for single, as well as multiple user systems. This study also provides extensive simulation results that investigate the calculation accuracy and speed advantages that Gooding’s technique possesses over that of the classic Monte-Carlo technique for BER estimation. These simulations also consider uncoded and coded cases, as well as single and multiple users. Finally, this study investigates the experimental real-time performance of the fully functional MU, adaptive coded, DS/SSMA communication system over varying channel conditions. During this part of the study, the channel conditions are varied over time, and the system’s adaptation (channel code switching) performance is observed through a real-time observation of the system’s estimated BER. This study also extends into cases with multiple system users. Since the adaptive coded system of this study does not require known data sequences (training sequences), inclusion of Gooding’s technique for real-time BER estimation through threshold modification and PER extrapolation in future 4G adaptive systems will enable better Quality-of-Service (QoS) management without sacrificing throughput. Furthermore, this study proves that when Gooding’s technique is applied to a coded system with a soft-output, it can be an effective technique for QoS monitoring, and should be considered in 4G systems of the future. / Dissertation (MEng (Computer Engineering))--University of Pretoria, 2007. / Electrical, Electronic and Computer Engineering / MEng / unrestricted