Design Of Area Delay Efficient Fixed-Point Lms Adaptive Filter For EEG Application

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INTERNATIONAL JOURNAL FOR TRENDS IN ENGINEERING & TECHNOLOGY VOLUME 5 ISSUE 1 – MAY 2015 - ISSN: 2349 - 9303

Design Of Area Delay Efficient Fixed-Point Lms Adaptive Filter For EEG Application R.Kanagarathinam, PG scholar, ECE Kalasalingam institute of technology santha.kanaga@gmail.com

R.Subhashini, Assistant Professor, ECE Kalasalingam institute of technology subhashini.rj@gmail.com

Abstract— An efficient architecture for the implementation of a delayed least mean square adaptive filter. A Novel partial product Generator is achieving lower adaptation-delay and Area delay consumption and propose a strategy for optimized balanced pipelining across the time-consuming combinational blocks of the structure. From synthesis results, the proposed design will offers less area-delay product (ADP) the best of the existing systolic structures, on average, for filter lengths N =8, 16, and 32. An efficient fixed-point implementation scheme of the proposed architecture, The EEG(electroencephalogram) is used for recording of electrical activity of the brain .During recording the EEG is contaminated by various artifacts as PLI(Power line interference), MA(Muscle artifact), EBA(Eye blink artifact). This paper gives Detail of various artifacts which occur in EEG signal. In this we study adaptive filter for reducing the EBA (eye blink artifact) noise from the EEG signal and to increase SNR (Signal to noise ratio).the analytical result matches with the simulation result is showed. Index Terms—Adaptive filters, circuit optimization, fixed-point arithmetic, least mean square (LMS) algorithms, EEG. found to be more efficient .The DLMS Adaptive filter proposed in Noise canceller application in EEG (Electro Encephalogram).

I. INTRODUCTION The least mean square (LMS) adaptive filter is the most popular and widely used adaptive filter, because involves a long critical path due to its inner-product computation to obtain the output from filter such that the critical path is required to be reduced by pipelined implementation when it exceeds to desired Sample Period of time. But the conventional LMS algorithm does not Support for pipelined implementation because of its recursive behavior, so they are modified to a form called the delayed LMS (DLMS) Algorithm, which allows pipelined implementation of the filter. A lot of work has been done to implement the DLMS algorithm in systolic architectures to increase the frequency but, they involve an adaptation delay for filter length N this is quite high for large order filters. We proposed a 2-bit multiplication cell, and with an efficient adder tree for pipelined inner-product computation to minimize the critical path and silicon area without increasing the number of adaptation delays. The existing work on the LMS adaptive filter does not discuss with the fixed-point implementation issues, such as the place of radix point, choose of word length, and quantization at various stages of computation. Therefore, fixed-point implementations in the proposed design reduce the number of pipeline delays along with the area and delay. The proposed design is

II. DELAYED LMS ALGORITHM For every input sample, the LMS algorithm calculates the filter output and finds the difference between the computed output and the desired response. Using this difference the filter weights are updated in each rotation. During the nth iteration LMS algorithm updates the weights as follows : Wn+1= Wn+ μ · e(n) · x(n)

(1a)

Where, E (n) = d (n) − y (n)

y(n) = W^Tn· x(n )

(1b)

Here, X (n) is the input vector W (n) is the weight vector of nth order LMS adaptive

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INTERNATIONAL JOURNAL FOR TRENDS IN ENGINEERING & TECHNOLOGY VOLUME 5 ISSUE 1 – MAY 2015 - ISSN: 2349 - 9303 filter at the nth iteration given by III. PROPOSED SYSTEM X (n) = [x (n), x (n − 1), · ·, x (n − N+1)] ^T In the conventional DLMS algorithm (Fig.1) the adaptation delay of m cycles amounts to the delay introduced by the whole of adaptive filter structure consisting of FIR filtering and weight adaptation process. But include, this adaptation delay could be decomposed into two concepts. One part is the delay introduced due to the Adaptive filtering and the other part is due to the delay involved in weight adaptation.

Wn= [wn (0), wn (1), · · ·, wn (N − 1)] ^T d (n) -desired response y(n) is the filter output of the nth iteration. e (n) is the error computed in the nth iteration which is used to update the weights the convergence-factor. The DLMS algorithm, instead of using the recentmost feedback-error e (n) corresponding to the nth iteration for updating the filter weights, it uses the delayed error e (n−m), (i.e.) the error corresponding to (n−m) the iteration for updating the current weight. The weight-update formula of DLMS algorithm is given by, Wn+1 = Wn + μ · e (n − m) · x(n − m)

(2)

Where, m is the adaptation-delay Fig 2: Structure of modified DLMS adaptive filter Based on such delay of decompositions, the structure of DLMS adaptive filter is shown in Fig.3. The modified adaptive filter architecture consists of two main computation blocks, first one is the error computation block and weight updatation block. The computation of Adaptive filter output and the final Subtraction to compute the feedback error is merged in the error computation unit to reduce the latency of error computation path. If the latency of computation of error is n1 cycles, the error computed by the structure at the nth cycle is e (n − n1), which is used with the input samples delayed by n1 cycles to generate the weight-raising term. The weight updating equation of the proposed delayed LMS algorithm is, Therefore given by, Fig .1. Structure of delayed LMS adaptive filter

Wn+1 = Wn + μ · e (n − n1) · x(n − n1) (3a)

The structure of conventional delayed LMS adaptive filter is shown in Figure. It can be seen that the adaptation delay m is the number of cycles required for the error corresponding to any given sampling instant to become available to the weight adaptation circuit.

Where, e (n − n1) = d(n − n1) − y(n − n1) And y (n) = W^T( n−n2) · x(n)

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(3b)

(3c)


INTERNATIONAL JOURNAL FOR TRENDS IN ENGINEERING & TECHNOLOGY VOLUME 5 ISSUE 1 – MAY 2015 - ISSN: 2349 - 9303

We can notice that during weight adaptation, the error with n1delays is used while the filtering unit uses the weights delayed by n2 cycles and this approach the adaptation-delay is effectively reduced by N2 cycles. The modified algorithm can be implemented efficiently with very low adaptationdelay which is not affected substantially by the increase in filter order.

V.WEIGHT-UPDATE BLOCK The proposed structure for the weightupdate block is shown in Fig. 8. It performs N multiply-accumulate operations of the form (μ × e) × xi + wi to update N filter weights. The step size μ is taken as a negative power of 2 to realize the multiplication with recently available error only by a shift operation each of the MAC units therefore performs the multiplication of the shifted value of error with the delayed input samples xi followed by the additions with the corresponding old weight values wi . All the N multiplications for the MAC operations are performed by N PPGs, followed by N shift add trees. Each of the PPGs creates L/2 partial products corresponding to the product of the recently shifted error value μ × e with L/2, the number of 2-bit digits of the input word xi, where the sub expression 3μ×e is shared within the multiplier. The final Result of MAC units constitute the desired updated weight to be used as inputs to the error computation block as well as the weight-update block for the next iteration.

IV.ERROR COMPUTATION BLOCK The proposed structure for errorcomputation unit of an N-tap DLMS adaptive filter is shown in Fig. 4. It consists of N number of 2-b partial product generators (PPG) corresponding to N multipliers and a cluster of L/2 binary adder trees, followed by a single shift–add tree. Each sub block is described in detail. The structure of each PPGis shown in Fig. It consists of L/2 number of 2-to-3 decoders and the same number of AND/OR cells (AOC).1 each of the 2-to-3 decoders takes a 2-b digit (u1u0) as input and produces three outputs b0 = u0 · . u1, b1 =. u0 · u1, and b2 = u0 · u1, such that b0 = 1 for (u1u0) =1, b1 = 1 for (u1u0) = 2, and b2 = 1 for (u1u0) =3. The decoder output b0, b1 and b2 along with w, 2w, and 3w are fall to an AOC, where w, 2w, and 3w are in 2‟s complement considerations and sign-extended to have (W + 2) bits each. To take care of the sign of the input samples while computing the partial product corresponding to the most significant digit (MSD), i.e., (uL−1uL−2) of the input sample, the AOC (L/2 − 1) is fed with w, −2w, and −w as input since (uL−1uL−2) can have four possible values 0, 1, −2, and −1.

Fig 3.Pipelined Structure of the Error-Computation Block

Fig 4.Proposed structure of the weight-update block.

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INTERNATIONAL JOURNAL FOR TRENDS IN ENGINEERING & TECHNOLOGY VOLUME 5 ISSUE 1 – MAY 2015 - ISSN: 2349 - 9303 VI. PROPOSED DLMS ADAPTIVE FILTER IN NOISE CANCELLER APPLICATION: EEG (Electro-Encephalogram):

VII.FIXED-POINT,SIMULATION ANALYSIS In this section, we discuss the fixed-point implementation and optimization of the proposed DLMS adaptive filter. A bit level pruning of the adder tree is also proposed to reduce the area

Electro-Encephalogram by adaptive filtering the eye forms an electric dipole. where the cornea is a positive and the retina is a negative. When the eye moves (saccade, blink or other movements), the electric field cover the eye the eye also changes, producing an electrical signal said as the electrooculogram (EOG). As this signal propagates over the scalp, it creates in the recorded electroencephalogram (EEG) as noise canceller or artifacts that present serious problems in EEG interpretation and analysis. There are at least two kinds of EOG artifact to be removed: those produced by the vertical eye movement (the corresponding EOG is called VEOG) and those produced by the horizontal eye movement (HEOG). Consequently, an EOG with two reference inputs is used in this application. Bellow Fig shows the EOG noise canceller. The basic input to the system is the EEG signal s (n). take up by a particular electrode. This signal is model as a combination of a true EEG x(n) and a noise component r(n), v(n) and v’(n) are the reference inputs, VEOG and HEOG,.

Fig.6. Fixed Point Representation of Binary Number A. Fixed-Point Design Considerations For fixed-point implementation, the choice of word lengths and radix points for input samples, weights, and internal signals need to be decided. Fig. 9 shows the fixed-point representation of a binary number. Let (X, Xi) be a fixed-point representation of a binary number where X is the word length and Xi is the integer length. The word length and location of radix point of xn and wn in Fig. 4 need to be predetermined by the hardware designer taking the design constraints, such as desired accuracy and hardware complexity, into consideration. Assuming (L, Li) and(W,Wi), respectively, as the representations of input signals and filter weights, all other signals can be decided as shown in The signal pi j , which is the output of PPG block has at most three times the value of input coefficients. Thus, we can add two more bits to the word length and to the integer length of the coefficients to avoid overflow.

B.Computer Simulation of the Proposed DLMS Filter The proposed fixed-point DLMS adaptive filter is used for system identification used. μ is set to 0.5, 0.25, and 0.125 for filter lengths 8, 16, and 32, respectively, such that the multiplication with μ does not require any additional circuits. For the fixed-point simulation, the word length and radix point of the input and coefficient are set to L = 16, Li = 2, W=16, Wi = 0, and the The fixed-point data type of all the other signals is obtained from. Each learning curve is averaged over 50 runs to obtain a clean curve. The proposed design was coded in C++ using System fixed-point library for different orders of the bandpass filter, that is, N = 8.

Fig.5.EOG Noise Cancller Respectively. V (n) and v’ (n) are correlated, in some unknown way, with the noise component r (n) in the primary input. The final output from the noise canceller e (n) is the rectified, or clean, EEG.

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INTERNATIONAL JOURNAL FOR TRENDS IN ENGINEERING & TECHNOLOGY VOLUME 5 ISSUE 1 – MAY 2015 - ISSN: 2349 - 9303 VIII.PERFOMANCE RESULTS If we consider each multiplier to have (L − 1) adders, then the existing designs involve 16N adders, while the propose done involves 10N +2 adders for L = 8. This section evaluates the performance of the proposed modified least mean square (LMS) algorithm and shows the simulation results. The first result declares about the output of LMS adaptive filter with delay. It is having some delay in the output of Delayed Least Mean Square adaptive filter. The result declares about the output of LMS adaptive filter without delay. After the clock input has given the output of the adaptive filter is achieved without delay. The Modelsim is the tool used here to check the performance of LMS adaptive filter. It is a complete HDL simulation environment that enables to verify the source code and functional and timing models using test bench

Fig.8. Final Noise Canceller using Adaptive Filter Output CONCLUSION We proposed an area–delay-power efficient low adaptation delay structure for fixed-point implementation of LMS adaptive filter. We used a new PPG for efficient implementation of general multiplications and inner-product computation by common sub expression sharing. Besides, we have proposed an efficient addition scheme for innerproduct computation to reduce the adaptation delay significantly in order to achieve faster convergence performance and to reduce the critical path to support high input-sampling rates. Aside from this, we proposed a strategy for optimized balanced pipelining across the time-consuming blocks of the structure to reduce the adaptation delay and power consumption, as well.

Fig.7. Output of Delayed LMS Adaptive Filter

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INTERNATIONAL JOURNAL FOR TRENDS IN ENGINEERING & TECHNOLOGY VOLUME 5 ISSUE 1 – MAY 2015 - ISSN: 2349 - 9303 The proposed structure involved significantly less adaptation delay and provided significant saving of ADP compared to the existing structures. We proposed a fixed-point implementation of the proposed architecture, and rectified the expression for steady-state error. We found that the steady-state MSE obtained from the analytical result matched well with the simulation result. We also discussed a pruning scheme that provides nearly 30% saving in the ADP, without a noticeable degradation of steady-state error performance.

REFERENCES [1] B. Widrow and S. D. Stearns, Adaptive Signal Processing., Englewood Cliffs,.NJ, USA: PrenticeHall, 1985. [2] S. Haykin and B. Widrow, Least-Mean-Square Adaptive Filters. Hoboken, NJ, USA: Wiley, 2003. [3] M. D. Meyer and D. P. Agrawal, “A modular pipelined implementation of a delayed LMS transversal adaptive filter,” in Proc., IEEE Int. Symp. Circuits Syst., May 1990, pp. 1943–1946. [4] G. Long, F. Ling, and J. G. Proakis, “The LMS algorithm with delayed coefficient adaptation,” IEEE Trans. Acoust., Speech, Signal Process. vol. 37, no. 9, pp. 1397–1405, Sep. 1989. , [5] G. Long, F. Ling, and J. G. Proakis, “Corrections to „The LMS algorithm with delayed coefficient adaptation‟,” IEEE Trans. Signal Process.vol. 40, no. 1, pp. 230–232, Jan. 1992. , [6] H. Herzberg and R. Haimi-Cohen, “A systolic array realization of an LMS adaptive filter and the effects of delayed adaptation,” . IEEE Trans Signal Process., vol. 40, no. 11, pp. 2799–2803, Nov. 1992. ,

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