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

Sparse nonlinear optimization for signal processing and communications

This dissertation proposes three classes of new sparse nonlinear optimization algorithms for network echo cancellation (NEC), 3-D synthetic aperture radar (SAR) image reconstruction, and adaptive turbo equalization in multiple-input multiple-output (MIMO) underwater acoustic (UWA) communications, respectively. For NEC, the proposed two proportionate affine projection sign algorithms (APSAs) utilize the sparse nature of the network impulse response (NIR). Benefiting from the characteristics of l₁-norm optimization, affine projection, and proportionate matrix, the new algorithms are more robust to impulsive interferences and colored input than the conventional adaptive algorithms. For 3-D SAR image reconstruction, the proposed two compressed sensing (CS) approaches exploit the sparse nature of the SAR holographic image. Combining CS with the range migration algorithms (RMAs), these approaches can decrease the load of data acquisition while recovering satisfactory 3-D SAR image through l₁-norm optimization. For MIMO UWA communications, a robust iterative channel estimation based minimum mean-square-error (MMSE) turbo equalizer is proposed for large MIMO detection. The MIMO channel estimation is performed jointly with the MMSE equalizer and the maximum a posteriori probability (MAP) decoder. The proposed MIMO detection scheme has been tested by experimental data and proved to be robust against tough MIMO channels. --Abstract, page iv

Unknown System Identification using LMS Algorithm

An adaptive filter is a digital filter that self adjusts its transfer function according to an optimizing algorithm which is most frequently Least Mean Square (LMS) algorithm. Due to the complexity of adaptive filtering most digital filters are FIR filter. There are numerous applications of adaptive filters like noise cancellations, echo cancellation, system modelling and identification, inverse system modelling, adaptive beam-forming etc. In this research article, adaptive LMS algorithm has been used for unknown system identification. The system identification is a category of adaptive filtering which find its numerous applications in diverse field like communication, image processing, speech processing etc

An FPGA architecture design of a high performance adaptive notch filter

The occurrence of narrowband interference near frequencies carrying information is a common problem in modern control and signal processing applications. A very narrow notch filter is required in order to remove the unwanted signal while not compromising the integrity of the carrier signal. In many practical situations, the interference may wander within a frequency band, in which case a wider notch filter would be needed to guarantee its removal, which may also allow for the degradation of information being carried in nearby frequencies. If the interference frequency could be autonomously tracked, a narrow bandwidth notch filter could be successfully implemented for the particular frequency. Adaptive signal processing is a powerful technique that can be used in the tracking and elimination of such a signal. An application where an adaptive notch filter becomes necessary is in biomedical instrumentation, such as the electrocardiogram recorder. The recordings can become useless when in the presence of electromagnetic fields generated by power lines. Research was conducted to fully characterize the interference. Research on notch filter structures and adaptive filter algorithms has been carried out. The lattice form filter structure was chosen for its inherent stability and performance benefits. A new adaptive filter algorithm was developed targeting a hardware implementation. The algorithm used techniques from several other algorithms that were found to be beneficial. This work developed the hardware implementation of a lattice form adaptive notch filter to be used for the removal of power line interference from electrocardiogram signals. The various design tradeo s encountered were documented. The final design was targeted toward multiple field programmable gate arrays using multiple optimization efforts. Those results were then compared. The adaptive notch filter was able to successfully track and remove the interfering signal. The lattice form structure utilized by the proposed filter was verified to exhibit an inherently stable realization. The filter was subjected to various environments that modeled the different power line disturbances that could be present. The final filter design resulted in a 3 dB bandwidth of 15.8908 Hz, and a null depth of 54 dB. For the baseline test case, the algorithm achieved convergence after 270 iterations. The final hardware implementation was successfully verified against the MATLAB simulation results. A speedup of 3.8 was seen between the Xilinx Virtex-5 and Spartan-II device technologies. The final design used a small fraction of the available resources for each of the two devices that were characterized. This would allow the component to be more readily available to be added to existing projects, or further optimized by utilizing additional logic

Design of a reusable distributed arithmetic filter and its application to the affine projection algorithm

Digital signal processing (DSP) is widely used in many applications spanning the spectrum from audio processing to image and video processing to radar and sonar processing. At the core of digital signal processing applications is the digital filter which are implemented in two ways, using either finite impulse response (FIR) filters or infinite impulse response (IIR) filters. The primary difference between FIR and IIR is that for FIR filters, the output is dependent only on the inputs, while for IIR filters the output is dependent on the inputs and the previous outputs. FIR filters also do not sur from stability issues stemming from the feedback of the output to the input that aect IIR filters. In this thesis, an architecture for FIR filtering based on distributed arithmetic is presented. The proposed architecture has the ability to implement large FIR filters using minimal hardware and at the same time is able to complete the FIR filtering operation in minimal amount of time and delay when compared to typical FIR filter implementations. The proposed architecture is then used to implement the fast affine projection adaptive algorithm, an algorithm that is typically used with large filter sizes. The fast affine projection algorithm has a high computational burden that limits the throughput, which in turn restricts the number of applications. However, using the proposed FIR filtering architecture, the limitations on throughput are removed. The implementation of the fast affine projection adaptive algorithm using distributed arithmetic is unique to this thesis. The constructed adaptive filter shares all the benefits of the proposed FIR filter: low hardware requirements, high speed, and minimal delay.Ph.D.Committee Chair: Anderson, Dr. David V.; Committee Member: Hasler, Dr. Paul E.; Committee Member: Mooney, Dr. Vincent J.; Committee Member: Taylor, Dr. David G.; Committee Member: Vuduc, Dr. Richar

Development of an Adaptive IIR Filter Based on Modified Robust Mixed-Norm Algorithm for Adaptive Noise Cancellation

Noise cancellation is one of the most important applications of adaptive filters. The employment of adaptive filtering in most digital signal processing tasks is currently an area of growing interest as adaptive filters, due to their dynamic nature, perform better than the traditional filters in compensating for random noise in their environment. However, the compensation for impulsive interference or noise is desired since most adaptive algorithms earlier proposed modelled noise as a random process of the White Gaussian distribution.  A modified robust mixed-norm (MRMN) algorithm recently proposed to compensate for impulsive interference has been found to be hardware efficient, however the MRMN algorithm has only been tested on adaptive FIR system identification task. In this paper, an adaptive IIR filter based on MRMN adaptive algorithm is proposed and tested for noise cancellation task. The developed filter structure was modelled and simulated in MATLAB environment. The results obtained showed that the MRMN algorithm does in fact compensate for the presence of impulsive interference, however, at a higher computational complexity relative to the LMS algorithm. Keywords: Noise cancellation, adaptive filtering, impulsive noise, adaptive algorithm, system identification, random noise DOI: 10.7176/CEIS/10-2-01 Publication date:March 31st 201

Reconfigurable Architecture for Noise Cancellation in Acoustic Environment Using Single Multiply Accumulate Adaline Filter

The creation of multiple applications with a higher level of complexity has been made possible by the usage of artificial neural networks (ANNs). In this research, an efficient flexible finite impulse response (FIR) filter structure called ADALINE (adaptive linear element) that makes use of a MAC (multiply accumulate) core is proposed. The least mean square (LMS) and recursive least square (RLS) algorithms are the most often used methods for maximizing filter coefficients. Despite outperforming the LMS, the RLS approach has not been favored for real-time applications due to its higher design arithmetic complexity. To achieve less computation, the fundamental filter has utilized an LMS-based tapping delay line filter, which is practically a workable option for an adaptive filtering algorithm. To discover the undiscovered system, the adjustable coefficient filters have been developed in the suggested work utilizing an optimal LMS approach. The 10-tap filter being considered here has been analyzed and synthesized utilizing field programmable gate array (FPGA) devices and programming in hardware description language. In terms of how well the resources were used, the placement and postrouting design performed well. If the implemented filter architecture is compared with the existing filter architecture, it reveals a 25% decrease in resources from the existing one and an increase in clock frequency of roughly 20%