A Digital Predistortion Scheme Exploiting Degrees-of-Freedom for Massive MIMO Systems

The primary source of nonlinear distortion in wireless transmitters is the power amplifier (PA). Conventional digital predistortion (DPD) schemes use high-order polynomials to accurately approximate and compensate for the nonlinearity of the PA. This is not practical for scaling to tens or hundreds of PAs in massive multiple-input multiple-output (MIMO) systems. There is more than one candidate precoding matrix in a massive MIMO system because of the excess degrees-of-freedom (DoFs), and each precoding matrix requires a different DPD polynomial order to compensate for the PA nonlinearity. This paper proposes a low-order DPD method achieved by exploiting massive DoFs of next-generation front ends. We propose a novel indirect learning structure which adapts the channel and PA distortion iteratively by cascading adaptive zero forcing precoding and DPD. Our solution uses a 3rd order polynomial to achieve the same performance as the conventional DPD using an 11th order polynomial for a 100x10 massive MIMO configuration. Experimental results show a 70% reduction in computational complexity, enabling ultra-low latency communications.Comment: IEEE International Conference on Communications 201

Adaptive Bayesian decision feedback equalizer for dispersive mobile radio channels

The paper investigates adaptive equalization of time dispersive mobile ratio fading channels and develops a robust high performance Bayesian decision feedback equalizer (DFE). The characteristics and implementation aspects of this Bayesian DFE are analyzed, and its performance is compared with those of the conventional symbol or fractional spaced DFE and the maximum likelihood sequence estimator (MLSE). In terms of computational complexity, the adaptive Bayesian DFE is slightly more complex than the conventional DFE but is much simpler than the adaptive MLSE. In terms of error rate in symbol detection, the adaptive Bayesian DFE outperforms the conventional DFE dramatically. Moreover, for severely fading multipath channels, the adaptive MLSE exhibits significant degradation from the theoretical optimal performance and becomes inferior to the adaptive Bayesian DFE

Subband decomposition techniques for adaptive channel equalisation

In this contribution, the convergence behaviour of the adaptive linear equaliser based on subband decomposition technique is investigated. Two different subband-based linear equalisers are employed, with the aim of improving the equaliser's convergence performance. Simulation results over three channel models having different spectral characteristic are presented. Computer simulations indicate that subband-based equalisers outperform the conventional fullband linear equaliser when channel exhibit severe spectral dynamic. Convergence rate of subband equalisers are governed by the slowest subband, whereby different convergence behaviour in each individual subband is observed. Finally, the complexity of fullband and subband equalisers is discussed

A performance comparison of fullband and different subband adaptive equalisers

We present two different fractionally spaced (FS) equalisers based on subband methods, with the aim of reducing the computational complexity and increasing the convergence rate of a standard fullband FS equaliser. This is achieved by operating in decimated subbands; at a considerably lower update rate and by exploiting the prewhitening effect that a filter bank has on the considerable spectral dynamics of a signal received through a severely distorting channel. The two presented subband structures differ in their level of realising the feedforward and feedback part of the equaliser in the subband domain, with distinct impacts on the updating. Simulation results pinpoint the faster convergence at lower cost for the proposed subband equalisers

Realization of Delayed Least Mean Square Adaptive Algorithm using Verilog HDL for EEG Signals

An efficient architecture for the implementation of delayed least mean square (DLMS) adaptive filter is presented in this paper. It is shown that the proposed architectures reduces the register complexity and also supports the faster convergence. Compared to transpose form, the direct form LMS adaptive filter has fast convergence but both has most similar critical path. Further it is shown that in most of the practical cases, very small adaptation delay is sufficient enough to implement a direct-form LMS adaptive filter where in normal cases a very high sampling rate is required and also it shows that no pipelining approach is necessary. From the above discussed estimations three different architectures of LMS adaptive filter has been designed. They are, first design comprise of zero delays i.e., with no adaptation delays, second design comprises of only single delay i.e., with only one adaptation delay, and lastly the third design comprises of two adaptation delays. Among all the three designs zero adaptation delay structure gives efficient performance comparatively. Design with zero adaptation delay involves the minimum energy per sample (EPS) and also minimum area compared to other two designs. The aim of this thesis is to design an efficient filter structures to create a system-on-chip (SoC) solution by using an optimized code for solving various adaptive filtering problems in the system. In this thesis our main focus is on interference cancellation in electroencephalogram (EEG) applications by using the proposed filter structures. Modern field programmable gate arrays (FPGAs) have the resources that are required to design an effective adaptive filtering structures. The designs are evaluated in terms of design time, area and delays