On the eigenfilter design method and its applications: a tutorial

The eigenfilter method for digital filter design involves the computation of filter coefficients as the eigenvector of an appropriate Hermitian matrix. Because of its low complexity as compared to other methods as well as its ability to incorporate various time and frequency-domain constraints easily, the eigenfilter method has been found to be very useful. In this paper, we present a review of the eigenfilter design method for a wide variety of filters, including linear-phase finite impulse response (FIR) filters, nonlinear-phase FIR filters, all-pass infinite impulse response (IIR) filters, arbitrary response IIR filters, and multidimensional filters. Also, we focus on applications of the eigenfilter method in multistage filter design, spectral/spacial beamforming, and in the design of channel-shortening equalizers for communications applications

Channel equalization to achieve high bit rates in discrete multitone systems

textMulticarrier modulation (MCM) techniques such as orthogonal frequency division multiplexing (OFDM) and discrete multi-tone (DMT) modulation are attractive for high-speed data communications due to the ease with which MCM can combat channel dispersion. With all the benefits MCM could give, DMT modulation has an extra ability to perform dynamic bit loading, which has the potential to exploit fully the available bandwidth in a slowly time-varying channel. In broadband wireline communications, DMT modulation is standardized for asymmetric digital subscribe line (ADSL) and very-high-bit-rate digital subscriber line (VDSL) modems. ADSL and VDSL standards are used by telephone companies to provide high speed data service to residences and offices. In an ADSL receiver, an equalizer is required to compensate for the channel’s dispersion in the time domain and the channel’s distortion in the frequency domain of the transmitted waveform. This dissertation proposes design methods for linear equalizers to increase the bit rate of the connection. The methods are amenable to implementation on programmable fixed-point digital signal processors, which are employed in ADSL/VDSL transceivers. A conventional ADSL equalizer consists of a time-domain equalizer, a fast Fourier transform, and a frequency domain equalizer. The time domain equalizer (TEQ) is a finite impulse response filter that when coupled with a discretized channel produces an equivalent channel whose impulse response is shorter than that of the discretized channel. This channel shortening is required by the ADSL standards. In this dissertation, I first propose a linear phase TEQ design that exploits symmetry in existing eigen-filter approaches such as minimum mean square error(MMSE), maximum shortening signal to noise ratio (MSSNR) and minimum intersymbol interference (Min-ISI) equalizers. TEQs with symmetric coefficients can reach the same performance as non-symmetric ones with much lower training complexity. Second, I improve Min-ISI design. I reformulate the cost function to make long TEQs design feasible. I remove the dependency of transmission delay in order to reduce the complexity associated with delay optimization. The quantized weighting is introduced to further lower the complexity. I also propose an iterative optimization procedure of Min-ISI that completely avoids Cholesky decomposition hence is better suited for a fixed-point implementation. Finally I propose a dual-path TEQ structure, which designs a standard singleFIR TEQ to achieve good bit rate over the entire transmission bandwidth, and designs another FIR TEQ to improve the bit rate over a subset of subcarriers. Dualpath TEQ can be viewed as a special case of a complex valued filter bank structure that delivers the best bit rate of existing DMT equalizers. However, dual-path TEQ provides a very good tradeoff between achievable bit rate vs. implementation complexity on a programmable digital signal processor.Electrical and Computer Engineerin

Fractional biorthogonal partners in channel equalization and signal interpolation

The concept of biorthogonal partners has been introduced recently by the authors. The work presented here is an extension of some of these results to the case where the upsampling and downsampling ratios are not integers but rational numbers, hence, the name fractional biorthogonal partners. The conditions for the existence of stable and of finite impulse response (FIR) fractional biorthogonal partners are derived. It is also shown that the FIR solutions (when they exist) are not unique. This property is further explored in one of the applications of fractional biorthogonal partners, namely, the fractionally spaced equalization in digital communications. The goal is to construct zero-forcing equalizers (ZFEs) that also combat the channel noise. The performance of these equalizers is assessed through computer simulations. Another application considered is the all-FIR interpolation technique with the minimum amount of oversampling required in the input signal. We also consider the extension of the least squares approximation problem to the setting of fractional biorthogonal partners

Design and implementation of low complexity wake-up receiver for underwater acoustic sensor networks

This thesis designs a low-complexity dual Pseudorandom Noise (PN) scheme for identity (ID) detection and coarse frame synchronization. The two PN sequences for a node are identical and are separated by a specified length of gap which serves as the ID of different sensor nodes. The dual PN sequences are short in length but are capable of combating severe underwater acoustic (UWA) multipath fading channels that exhibit time varying impulse responses up to 100 taps. The receiver ID detection is implemented on a microcontroller MSP430F5529 by calculating the correlation between the two segments of the PN sequence with the specified separation gap. When the gap length is matched, the correlator outputs a peak which triggers the wake-up enable. The time index of the correlator peak is used as the coarse synchronization of the data frame. The correlator is implemented by an iterative algorithm that uses only one multiplication and two additions for each sample input regardless of the length of the PN sequence, thus achieving low computational complexity. The real-time processing requirement is also met via direct memory access (DMA) and two circular buffers to accelerate data transfer between the peripherals and the memory. The proposed dual PN detection scheme has been successfully tested by simulated fading channels and real-world measured channels. The results show that, in long multipath channels with more than 60 taps, the proposed scheme achieves high detection rate and low false alarm rate using maximal-length sequences as short as 31 bits to 127 bits, therefore it is suitable as a low-power wake-up receiver. The future research will integrate the wake-up receiver with Digital Signal Processors (DSP) for payload detection. --Abstract, page iv

Doctor of Philosophy

dissertationThe demand for high speed communication has been increasing in the past two decades. Multicarrier communication technology has been suggested to address this demand. Orthogonal frequency-division multiplexing (OFDM) is the most widely used multicarrier technique. However, OFDM has a number of disadvantages in time-varying channels, multiple access, and cognitive radios. On the other hand, filterbank multicarrier (FBMC) communication has been suggested as an alternative to OFDM that can overcome the disadvantages of OFDM. In this dissertation, we investigate the application of filtered multitone (FMT), a subset of FBMC modulation methods, to slow fading and fast fading channels. We investigate the FMT transmitter and receiver in continuous and discrete time domains. An efficient implementation of FMT systems is derived and the conditions for perfect reconstruction in an FBMC communication system are presented. We derive equations for FMT in slow fading channels that allow evaluation of FMT when applied to mobile wireless communication systems. We consider using fractionally spaced per tone channel equalizers with different number of taps. The numerical results are presented to investigate the performance of these equalizers. The numerical results show that single-tap equalizers suffice for typical wireless channels. The equalizer design study is advanced by introducing adaptive equalizers which use channel estimation. We derive equations for a minimum mean square error (MMSE) channel estimator and improve the channel estimation by considering the finite duration of channel impulse response. The results of optimum equalizers (when channel is known perfectly) are compared with those of the adaptive equalizers, and it is found that a loss of 1 dB or less incurs. We also introduce a new form of FMT which is specially designed to handle doubly dispersive channels. This method is called FMT-dd (FMT for doubly dispersive channels). The proposed FMT-dd is applied to two common methods of data symbol orientation in the time-frequency space grid; namely, rectangular and hexagonal lattices. The performance of these methods along with OFDM and the conventional FMT are compared and a significant improvement in performance is observed. The FMT-dd design is applied to real-world underwater acoustic (UWA) communication channels. The experimental results from an at-sea experiment (ACOMM10) show that this new design provides a significant gain over OFDM. The feasibility of implementing a MIMO system for multicarrier UWA communication channels is studied through computer simulations. Our study emphasizes the bandwidth efficiency of multicarrier MIMO communications .We show that the value of MIMO to UWA communication is very limited

Underwater acoustic communication under doppler effects

In this thesis we perform a research survey of the three available technologies for wireless underwater communications. We discuss the main features and drawbacks inherent to acoustic, RF, and optical communications. We focus our research on underwater acoustic communications, and we analyze and evaluate the channel frequency response of Arraial do Cabo using data acquired in situ. We further investigate the Doppler effect, a phenomenon that is inherent to underwater acoustic channels. We analyze and justify a compensation algorithm to mitigate the Doppler effects. We propose a simplified algorithm version for minimizing the required number of pilot symbols. We also develop a simple strategy to determine how often our proposed compensation method should be retrained. Our main contribution is the proposal of a new receiver design to deal with Doppler effects. We present the idea of iteratively adapt the correlator filter placed at the receiver side. We show that the adaptation of this filter’s support reduces the inter-symbol interference of the estimated symbols. Besides this idea, we demonstrate that the time-dependent phase-shift component of the received signal should be removed beforehand. That is, we propose a modification in the signal processing sequence blocks for improving the symbol estimation. For testing and comparing this new receiver design, we implement a communication model encompassing physical layer aspects. We perform several numerical simulations for single-carrier and multicarrier systems. Simulation results show that our proposal might provide a reduction in the bit error rate for high signal-to-noise ratios. This performance improvement can be observed for all tested relative movement, and even with dense digital signal constellation.Nesta tese foi realizada uma pesquisa extensa sobre as tecnologias existentes para comunicação sem fio subaquática. Foram analisadas as principais características das comunicações acústicas, RF e ótica. O estudo foi aprofundado na comunicação acústica, e foi realizada uma análise da resposta em frequência do canal de Arraial do Cabo com dados adquiridos no local. O efeito Doppler, um fenômeno inerente aos canais subaquáticos acústicos, foi investigado de forma minuciosa. Dentre as técnicas estudas para compensação deste efeito, foi escolhido um algoritmo adaptativo, o qual foi re-analisado com uma nova abordagem. Uma versão simplificada deste algoritmo foi proposta para reduzir a quantidade de símbolos pilotos. Foi também desenvolvida uma estratégia para determinar a frequência de treinamento deste novo algoritmo. A principal contribuição da tese é a proposta de uma nova estrutura de receptor para compensar o efeito Doppler. Nesta estrutura, é proposta a adaptação de forma iterativa do filtro correlator. A adaptação do suporte temporal deste filtro reduz a interferência inter-simbólica. Além desta ideia, foi demonstrado que a componente de fase do sinal recebido, que é dependente do tempo, deve ser removida em um estágio anterior ao usual. Ou seja, foi proposta uma modificação na sequência do processamento do sinal recebido para melhorar a sua estimativa. Para testar esta nova estrutura do receptor, foi implementado um sistema de comunicação. Foram realizadas simulações numéricas com sistemas de uma única e de múltiplas portadoras. Os resultados das simulações mostram que a nova estrutura pode reduzir a quantidade de erros de bits para altos valores de razão sinal-ruído. A melhora do desempenho pode ser observada em todas as velocidades relativas testadas, e também para constelações densas