VLSI Design and Implementation for Adaptive Filter using LMS Algorithm

Adaptive filters, as part of digital signal systems, have been widely used, as well as in applications such as adaptive noise cancellation, adaptive beam forming, channel equalization, and system identification. However, its implementation takes a great deal and becomes a very important field in digital system world. When FPGA (Field Programmable Logic Array) grows in area and provides a lot of facilities to the designers, it becomes an important competitor in the signal processing market. In general FIR structure has been used more successfully than IIR structure in adaptive filters. However, when the adaptive FIR filter was made this required appropriate algorithm to update the filter’s coefficients. The algorithm used to update the filter coefficient is the Least Mean Square (LMS) algorithm which is known for its simplification, low computational complexity, and better performance in different running environments. When compared to other algorithms used for implementing adaptive filters the LMS algorithm is seen to perform very well in terms of the number of iterations required for convergence. This phenomenon can be achieved by a sufficient choice of bit length to represent the filter’s coefficients. This paper presents a lowcost and high performance programmable digital finite impulse response (FIR) filter. It follows the adaptive algorithm used for the development of the system. The architecture employs the computation sharing algorithm to reduce the computation complexity

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

IMPLEMENTATION OF NOISE CANCELLATION WITH HARDWARE DESCRIPTION LANGUAGE

The objective of this project is to implement noise cancellation technique on an FPGA using Hardware Description Language. The performance of several adaptive algorithms is compared to determine the desirable algorithm used for adaptive noise cancellation system. The project will focus on the implementation of adaptive filter with least-meansquares (LMS) algorithm or normalized least-mean-squares (NLMS) algorithm to cancel acoustic noises. This noise consists of extraneous or unwanted waveforms that can interfere with communication. Due to the simplicity and effectiveness of adaptive noise cancellation technique, it is used to remove the noise component from the desired signal. The project is divided into four main parts: research, Matlab simulation, ModelSim simulation and hardware implementation. The project starts with research on several noise cancellation techniques, and then with Matlab code, Simulink and FDA tool, the adaptive noise cancellation system is designed with the implementation of the LMS algorithm, NLMS algorithm and recursive-least-square algorithm to remove the interference noise. By using the Matlab code and Simulink, the noise that interfered with a sinusoidal signal and a record of music can be removed. The original signal in turns can be retrieved from the noise corrupted signal by changing the coefficient of the filter. Since filter is the important component in adaptive filtering process, the filter is designed first before adding adaptive algorithm. A Finite Impulse Response (FIR) filter is designed and the desired result of functional simulation and timing simulation is obtained through ModelSim and Integrated Software Environment (ISE) software and FPGA implementation. Finally the adaptive algorithm is added to the filter, and implemented in the FPGA. The noise is greatly reduced in Matlab simulation, functional simulation and timing simulation. Hence the results of this project show that noise cancellation with adaptive filter is feasible

REAL-TIME ADAPTIVE PULSE COMPRESSION ON RECONFIGURABLE, SYSTEM-ON-CHIP (SOC) PLATFORMS

New radar applications need to perform complex algorithms and process a large quantity of data to generate useful information for the users. This situation has motivated the search for better processing solutions that include low-power high-performance processors, efficient algorithms, and high-speed interfaces. In this work, hardware implementation of adaptive pulse compression algorithms for real-time transceiver optimization is presented, and is based on a System-on-Chip architecture for reconfigurable hardware devices. This study also evaluates the performance of dedicated coprocessors as hardware accelerator units to speed up and improve the computation of computing-intensive tasks such matrix multiplication and matrix inversion, which are essential units to solve the covariance matrix. The tradeoffs between latency and hardware utilization are also presented. Moreover, the system architecture takes advantage of the embedded processor, which is interconnected with the logic resources through high-performance buses, to perform floating-point operations, control the processing blocks, and communicate with an external PC through a customized software interface. The overall system functionality is demonstrated and tested for real-time operations using a Ku-band testbed together with a low-cost channel emulator for different types of waveforms

Analysis of the sign regressor least mean fourth adaptive algorithm

A novel algorithm, called the signed regressor least mean fourth (SRLMF) adaptive algorithm, that reduces the computational cost and complexity while maintaining good performance is presented. Expressions are derived for the steady-state excess-mean-square error (EMSE) of the SRLMF algorithm in a stationary environment. A sufficient condition for the convergence in the mean of the SRLMF algorithm is derived. Also, expressions are obtained for the tracking EMSE of the SRLMF algorithm in a nonstationary environment, and consequently an optimum value of the step-size is obtained. Moreover, the weighted variance relation has been extended in order to derive expressions for the mean-square error (MSE) and the mean-square deviation (MSD) of the proposed algorithm during the transient phase. Computer simulations are carried out to corroborate the theoretical findings. It is shown that there is a good match between the theoretical and simulated results. It is also shown that the SRLMF algorithm has no performance degradation when compared with the least mean fourth (LMF) algorithm. The results in this study emphasize the usefulness of this algorithm in applications requiring reduced implementation costs for which the LMF algorithm is too complex

Low Power Adaptive Equaliser Architectures for Wireless LMMSE Receivers

Power consumption requires critical consideration during system design for portable wireless communication devices as it has a direct influence on the battery weight and volume required for operation. Wideband Code Division Multiple Access (W-CDMA) techniques are favoured for use in future generation mobile communication systems. This thesis investigates novel low power techniques for use in system blocks within a W-CDMA adaptive linear minimum mean squared error (LMMSE) receiver architecture. Two low power techniques are presented for reducing power dissipation in the LMS adaptive filter, this being the main power consuming block within this receiver. These low power techniques are namely the decorrelating transform, this is a differential coefficient technique, and the variable length update algorithm which is a dynamic tap-length optimisation technique. The decorrelating transform is based on the principle of reducing the wordlength of filter coefficients by using the computed difference between adjacent coefficients in calculation of the filter output. The effect of reducing the wordlength of filter coefficients being presented to multipliers in the filter is a reduction in switching activity within the multiplier thus reducing power consumed. In the case of the LMS adaptive filter, with coefficients being continuously updated, the decorrelating transform is applied to these calculated coefficients with minimal hardware or computational overhead. The correlation between filter coefficients is exploited to achieve a wordlength reduction from 16 bits down to 10 bits in the FIR filter block. The variable length update algorithm is based on the principle of optimising the number of operational filter taps in the LMS adaptive filter according to operating conditions. The number of taps in operation can be increased or decreased dynamically according to the mean squared error at the output of the filter. This algorithm is used to exploit the fact that when the SNR in the channel is low the minimum mean squared error of the short equaliser is almost the same as that of the longer equaliser. Therefore, minimising the length of the equaliser will not result in poorer MSE performance and there is no disadvantage in having fewer taps in operation. If fewer taps are in operation then switching will not only be reduced in the arithmetic blocks but also in the memory blocks required by the LMS algorithm and FIR filter process. This reduces the power consumed by both these computation intensive functional blocks. Power results are obtained for equaliser lengths from 73 to 16 taps and for operation with varying input SNR. This thesis then proposes that the variable length LMS adaptive filter is applied in the adaptive LMMSE receiver to create a low power implementation. Power consumption in the receiver is reduced by the dynamic optimisation of the LMS receiver coefficient calculation. A considerable power saving is seen to be achieved when moving from a fixed length LMS implementation to the variable length design. All design architectures are coded in Verilog hardware description language at register transfer level (RTL). Once functional specification of the design is verified, synthesis is carried out using either Synopsys DesignCompiler or Cadence BuildGates to create a gate level netlist. Power consumption results are determined at the gate level and estimated using the Synopsys DesignPower tool

The design and implementation of a microprocessor controlled adaptive filter

This thesis describes the construction and implementation of a microprocessor controlled recursive adaptive filter applied as a noise canceller. It describes the concept of the adaptive noise canceller, a method of estimating the received signal corrupted with additive interference (noise). This canceller has two inputs, the primary input containing the corrupted signal and the reference input consisting of the additive noise correlated in some unknown way to the primary noise. The reference input is filtered and subtracted from the primary input without degrading the desired components of the signal. This filtering process is adaptive and based on Widrow-Hoff Least-Mean-Square algorithm. Adaptive filters are programmable and have the capability to adjust their own parameters in situations where minimum piori knowledge is available about the inputs. For recursive filters, these parameters include feed-forward (non-recursive) as well as feedback (recursive) coefficients. A new design and implementation of the adaptive filter is suggested which uses a high speed 68000 microprocessor to accomplish the coefficients updating operation. Many practical problems arising in the hardware implementation are investigated. Simulation results illustrate the ability of the adaptive noise canceller to have an acceptable performance when the coefficients updating operation is carried out once every N sampling periods. Both simulation and hardware experimental results are in agreement

Optimisation of multiplier-less FIR filter design techniques

This thesis is concerned with the design of multiplier-less (ML) finite impulse response (FIR) digital filters. The use of multiplier-less digital filters results in simplified filtering structures, better throughput rates and higher speed. These characteristics are very desirable in many DSP systems. This thesis concentrates on the design of digital filters with power-of-two coefficients that result in simplified filtering structures. Two distinct classesof ML FIR filter design algorithms are developed and compared with traditional techniques. The first class is based on the sensitivity of filter coefficients to rounding to power-of-two. Novel elements include extending of the algorithm for multiple-bands filters and introducing mean square error as the sensitivity criterion. This improves the performance of the algorithm and reduces the complexity of resulting filtering structures. The second class of filter design algorithms is based on evolutionary techniques, primarily genetic algorithms. Three different algorithms based on genetic algorithm kernel are developed. They include simple genetic algorithm, knowledge-based genetic algorithm and hybrid of genetic algorithm and simulated annealing. Inclusion of the additional knowledge has been found very useful when re-designing filters or refining previous designs. Hybrid techniques are useful when exploring large, N-dimensional searching spaces. Here, the genetic algorithm is used to explore searching space rapidly, followed by fine search using simulated annealing. This approach has been found beneficial for design of high-order filters. Finally, a formula for estimation of the filter length from its specification and complementing both classes of design algorithms, has been evolved using techniques of symbolic regression and genetic programming. Although the evolved formula is very complex and not easily understandable, statistical analysis has shown that it produces more accurate results than traditional Kaiser's formula. In summary, several novel algorithms for the design of multiplier-less digital filters have been developed. They outperform traditional techniques that are used for the design of ML FIR filters and hence contributed to the knowledge in the field of ML FIR filter design