55 research outputs found

    Viterbi algorithm in continuous-phase frequency shift keying

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    The Viterbi algorithm, an application of dynamic programming, is widely used for estimation and detection problems in digital communications and signal processing. It is used to detect signals in communication channels with memory, and to decode sequential error-control codes that are used to enhance the performance of digital communication systems. The Viterbi algorithm is also used in speech and character recognition tasks where the speech signals or characters are modeled by hidden Markov models. This project explains the basics of the Viterbi algorithm as applied to systems in digital communication systems, and speech and character recognition. It also focuses on the operations and the practical memory requirements to implement the Viterbi algorithm in real-time. A forward error correction technique known as convolutional coding with Viterbi decoding was explored. In this project, the basic Viterbi decoder behavior model was built and simulated. The convolutional encoder, BPSK and AWGN channel were implemented in MATLAB code. The BER was tested to evaluate the decoding performance. The theory of Viterbi Algorithm is introduced based on convolutional coding. The application of Viterbi Algorithm in the Continuous-Phase Frequency Shift Keying (CPFSK) is presented. Analysis for the performance is made and compared with the conventional coherent estimator. The main issue of this thesis is to implement the RTL level model of Viterbi decoder. The RTL Viterbi decoder model includes the Branch Metric block, the Add-Compare-Select block, the trace-back block, the decoding block and next state block. With all done, we further understand about the Viterbi decoding algorithm

    Efficient human-machine control with asymmetric marginal reliability input devices

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    Input devices such as motor-imagery brain-computer interfaces (BCIs) are often unreliable. In theory, channel coding can be used in the human-machine loop to robustly encapsulate intention through noisy input devices but standard feedforward error correction codes cannot be practically applied. We present a practical and general probabilistic user interface for binary input devices with very high noise levels. Our approach allows any level of robustness to be achieved, regardless of noise level, where reliable feedback such as a visual display is available. In particular, we show efficient zooming interfaces based on feedback channel codes for two-class binary problems with noise levels characteristic of modalities such as motor-imagery based BCI, with accuracy <75%. We outline general principles based on separating channel, line and source coding in human-machine loop design. We develop a novel selection mechanism which can achieve arbitrarily reliable selection with a noisy two-state button. We show automatic online adaptation to changing channel statistics, and operation without precise calibration of error rates. A range of visualisations are used to construct user interfaces which implicitly code for these channels in a way that it is transparent to users. We validate our approach with a set of Monte Carlo simulations, and empirical results from a human-in-the-loop experiment showing the approach operates effectively at 50-70% of the theoretical optimum across a range of channel conditions

    Iterative receiver in multiuser relaying systems with fast frequency-hopping modulation

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    In this thesis, a novel iterative receiver and its improved version are proposed for relay-assisted multiuser communications, in which multiple users transmit to a destination with the help of a relay and using fast frequency-hopping modulation. Each user employs a channel encoder to protect its information and facilitate interference cancellation at the receiver. The signal received at the relay is either amplified, or partially decoded with a simple energy detector, before being forwarded to the destination. Under flat Rayleigh fading channels, the receiver at the destination can be implemented non-coherently, i.e., it does not require the instantaneous channel information to demodulate the users’ transmitted signals. The proposed iterative algorithm at the destination exploits the soft outputs of the channel decoders to successively extract the maximum likelihood symbols of the users and perform interference cancellation. The iterative method is successfully applied for both cases of amplify-and-forward and partial decode-and-forward relaying. The error performance of the proposed iterative receiver is investigated by computer simulation. Under the same spectral efficiency, simulation results demonstrate the excellent performance of the proposed receiver when compared to the performance of decoding without interference cancellation as well as the performance of the maximum likelihood multiuser detection previously developed for uncoded transmission. Simulation results also suggest that a proper selection of channel coding schemes can help to support significant more users without consuming extra system resources. In addition, to further enhance the receiver’s performance in terms of the bit error rate, an improved version of the iterative receiver is presented. Such an improved receiver invokes inner-loop iterations between the channel decoders and the demappers in such a way that the soft outputs of the channel decoders are also used to refine the outputs of the demappers for every outer-loop iteration. Simulation results indicate a performance gain of about 2.5dB by using the two-loop receiver when compared to the performance of the first proposed receiver

    Transmission Experiment of Bandwidth Compressed Carrier Aggregation in a Realistic Fading Channel

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    In this paper, an experimental testbed is designed to evaluate the performance of a bandwidth compressed multicarrier technique termed spectrally efficient frequency division multiplexing (SEFDM) in a carrier aggregation (CA) scenario1. Unlike orthogonal frequency division multiplexing (OFDM), SEFDM is a non-orthogonal waveform which, relative to OFDM, packs more sub-carriers in a given bandwidth, thereby improving spectral efficiency. CA is a long term evolution-advanced (LTE-Advanced) featured technique that offers a higher throughput by aggregating multiple legacy radio bands. Considering the scarcity of radio spectrum, SEFDM signals can be utilized to enhance CA performance. The combination of the two techniques results in a larger number of aggregated component carriers (CCs) and therefore increased data rate in a given bandwidth with no additional spectral allocation. It is experimentally shown that CA-SEFDM can aggregate up to 7 CCs in a limited bandwidth while CA-OFDM can only put 5 CCs in the same bandwidth. In this work, LTE-like framed CA-SEFDM signals are generated and delivered through a realistic LTE channel. A complete experimental setup is described together with error performance and effective spectral efficiency comparisons. Experimental results show that the measured BER performance for CA-SEFDM is very close to CA-OFDM and the effective spectral efficiency of CA-SEFDM can be substantially higher than that of CA-OFDM

    Codes on Graphs and More

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    Modern communication systems strive to achieve reliable and efficient information transmission and storage with affordable complexity. Hence, efficient low-complexity channel codes providing low probabilities for erroneous receptions are needed. Interpreting codes as graphs and graphs as codes opens new perspectives for constructing such channel codes. Low-density parity-check (LDPC) codes are one of the most recent examples of codes defined on graphs, providing a better bit error probability than other block codes, given the same decoding complexity. After an introduction to coding theory, different graphical representations for channel codes are reviewed. Based on ideas from graph theory, new algorithms are introduced to iteratively search for LDPC block codes with large girth and to determine their minimum distance. In particular, new LDPC block codes of different rates and with girth up to 24 are presented. Woven convolutional codes are introduced as a generalization of graph-based codes and an asymptotic bound on their free distance, namely, the Costello lower bound, is proven. Moreover, promising examples of woven convolutional codes are given, including a rate 5/20 code with overall constraint length 67 and free distance 120. The remaining part of this dissertation focuses on basic properties of convolutional codes. First, a recurrent equation to determine a closed form expression of the exact decoding bit error probability for convolutional codes is presented. The obtained closed form expression is evaluated for various realizations of encoders, including rate 1/2 and 2/3 encoders, of as many as 16 states. Moreover, MacWilliams-type identities are revisited and a recursion for sequences of spectra of truncated as well as tailbitten convolutional codes and their duals is derived. Finally, the dissertation is concluded with exhaustive searches for convolutional codes of various rates with either optimum free distance or optimum distance profile, extending previously published results

    The Deep Space Network: A Radio Communications Instrument for Deep Space Exploration

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    The primary purpose of the Deep Space Network (DSN) is to serve as a communications instrument for deep space exploration, providing communications between the spacecraft and the ground facilities. The uplink communications channel provides instructions or commands to the spacecraft. The downlink communications channel provides command verification and spacecraft engineering and science instrument payload data

    Performance Prediction of a Turbo-coded Link in Fading Channels

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    Channel coding is the method of adding redundancy to the data in order to reduce the frequency of errors or to increase the capacity of a channel. Turbo codes are the most superior class of codes making achievable channel capacity almost at par with the Shannon limits. In Adaptive Modulation and Coding (AMC) the prediction of error performance of a channel is an important step before choosing one of the Modulation Coding Scheme (MCS). Since in Turbo-coded system we donot have analytical relations to relate error performance with the Signal to Noise Ratio (SNR). Therefore, normally simulation results are stored in the form of the look up tables. In this work we propose an error performance prediction model for a BPSK modulated Turbo-coded link. This model predicts performance addressing the fading phenomena for wireless radio channels. It takes the large variations in the SNR level within a code block into account along with the coding parameters. The SNR dB values profile inside a code block is considered in terms of their mean and variance. The model proposed is UMTS compliant and is continuous for values of the MCS code rate and the mean and variance of SNR dB values. It is an easier way of predicting link level performance as it replaces the discrete look up tables. Unlike the look-up tables it can be used for differentiation based analytical techniques used in system level optimization

    On Coding and Detection Techniques for Two-Dimensional Magnetic Recording

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    Edited version embargoed until 15.04.2020 Full version: Access restricted permanently due to 3rd party copyright restrictions. Restriction set on 15/04/2019 by AS, Doctoral CollegeThe areal density growth of magnetic recording systems is fast approaching the superparamagnetic limit for conventional magnetic disks. This is due to the increasing demand for high data storage capacity. Two-dimensional Magnetic Recording (TDMR) is a new technology aimed at increasing the areal density of magnetic recording systems beyond the limit of current disk technology using conventional disk media. However, it relies on advanced coding and signal processing techniques to achieve areal density gains. Current state of the art signal processing for TDMR channel employed iterative decoding with Low Density Parity Check (LDPC) codes, coupled with 2D equalisers and full 2D Maximum Likelihood (ML) detectors. The shortcoming of these algorithms is their computation complexity especially with regards to the ML detectors which is exponential with respect to the number of bits involved. Therefore, robust low-complexity coding, equalisation and detection algorithms are crucial for successful future deployment of the TDMR scheme. This present work is aimed at finding efficient and low-complexity coding, equalisation, detection and decoding techniques for improving the performance of TDMR channel and magnetic recording channel in general. A forward error correction (FEC) scheme of two concatenated single parity bit systems along track separated by an interleaver has been presented for channel with perpendicular magnetic recording (PMR) media. Joint detection decoding algorithm using constrained MAP detector for simultaneous detection and decoding of data with single parity bit system has been proposed. It is shown that using the proposed FEC scheme with the constrained MAP detector/decoder can achieve a gain of up to 3dB over un-coded MAP decoder for 1D interference channel. A further gain of 1.5 dB was achieved by concatenating two interleavers with extra parity bit when data density along track is high. The use of single bit parity code as a run length limited code as well as an error correction code is demonstrated to simplify detection complexity and improve system performance. A low-complexity 2D detection technique for TDMR system with Shingled Magnetic Recording Media (SMR) was also proposed. The technique used the concatenation of 2D MAP detector along track with regular MAP detector across tracks to reduce the complexity order of using full 2D detection from exponential to linear. It is shown that using this technique can improve track density with limited complexity. Two methods of FEC for TDMR channel using two single parity bit systems have been discussed. One using two concatenated single parity bits along track only, separated by a Dithered Relative Prime (DRP) interleaver and the other use the single parity bits in both directions without the DRP interleaver. Consequent to the FEC coding on the channel, a 2D multi-track MAP joint detector decoder has been proposed for simultaneous detection and decoding of the coded single parity bit data. A gain of up to 5dB was achieved using the FEC scheme with the 2D multi-track MAP joint detector decoder over un-coded 2D multi-track MAP detector in TDMR channel. In a situation with high density in both directions, it is shown that FEC coding using two concatenated single parity bits along track separated by DRP interleaver performed better than when the single parity bits are used in both directions without the DRP interleaver.9mobile Nigeri

    Applications of microprocessors in digital high frequency radio communications

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    This thesis describes the application of VLSI devices to channel evaluation and communication techniques over ionospheric radio paths. Digital signal processing techniques using microprocessors and charge coupled devices are described in detail. A novel method for observing interference and fading patterns on HF channels is described. Error control coding schemes and digital modulation techniques are combined in a design for an adaptive modem for use over HF radio links. Results of narrow-band interference measurements, error patterns and coding performance are presented

    Polar-Coded OFDM with Index Modulation

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    Polar codes, as the first error-correcting codes with an explicit construction to provably achieve thesymmetric capacity of memoryless channels, which are constructed based on channel polarization, have recently become a primary contender in communication networks for achieving tighter requirements with relatively low complexity. As one of the contributions in this thesis, three modified polar decoding schemes are proposed. These schemes include enhanced versions of successive cancellation-flip (SC-F), belief propagation (BP), and sphere decoding (SD). The proposed SC-F utilizes novel potential incorrect bits selection criteria and stack to improve its error correction performance. Next, to make the decoding performance of BP better, permutation and feedback structure are utilized. Then, in order to reduce the complexity without compromising performance, a SD by using novel decoding strategies according to modified path metric (PM) and radius extension is proposed. Additionally, to solve the problem that BP has redundant iterations, a new stopping criterion based on bit different ratio (BDR) is proposed. According to the simulation results and mathematical proof, all proposed schemes can achieve corresponding performance improvement or complexity reduction compared with existing works. Beside applying polar coding, to achieve a reliable and flexible transmission in a wireless communication system, a modified version of orthogonal frequency division multiplexing (OFDM) modulation based on index modulation, called OFDM-in-phase/quadrature-IM (OFDM-I/Q-IM), is applied. This modulation scheme can simultaneously improve spectral efficiency and bit-error rate (BER) performance with great flexibility in design and implementation. Hence, OFDM-I/Q-IM is considered as a potential candidate in the new generation of cellular networks. As the main contribution in this work, a polar-coded OFDM-I/Q-IM system is proposed. The general design guidelines for overcoming the difficulties associated with the application of polar codes in OFDM-I/Q-IM are presented. In the proposed system, at the transmitter, we employ a random frozen bits appending scheme which not only makes the polar code compatible with OFDM-I/Q-IM but also improves the BER performance of the system. Furthermore, at the receiver, it is shown that the \textit{a posteriori} information for each index provided by the index detector is essential for the iterative decoding of polar codes by the BP algorithm. Simulation results show that the proposed polar-coded OFDM-I/Q-IM system outperforms its OFDM counterpart in terms of BER performance
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