Adaptive Filters Introduction Adaptive filters are considered nonlinear system therefore their behavior analysis is more complicated

than for fixed filters. On the other hand, because the adaptive filters are self designing filters, from the practitioners point of view their design can be considered less involved than in the case of digital filters with fixed coefficients. The general set up of an adaptive filtering environment is illustrated in Fig. 1.1, where k is the Iteration number, x(k) denotes the input signal, y(k) is the adaptive filter output signal, and d(k) Defines the desired signal. The error signal e(k) is calculated as d(k) y(k). The error signal is then used to form a performance (or objective) function that is required by the adaptation algorithm in order to determine the appropriate updating of the filter coefficients. The minimization of the objective function implies that the adaptive filter output signal is matching the desired signal in some sense.

Fig (a) adaptive-filter configuration

then used to form a performance (or objective) function that is required by the adaptation algorithm in order to determine the appropriate updating of the filter coefficients. The minimization of the objective function implies that the adaptive-filter output signal is matching the desired signal in some sense. The complete specification of an adaptive system, as shown in Fig. 1.1, consists of three items 1) Application: The type of application is defined by the choice of the signals acquired from the Environment to be the input and desired output signals. The number of different applications in Which adaptive techniques are being successfully used has increased enormously during the last Two decades. Some examples are echo cancellation, equalization of dispersive channels, system Identification, signal enhancement, adaptive beam forming, noise cancelling, and control the study of different applications. However, some applications are considered in some detail.

2) Adaptive-Filter Structure: The adaptive filter can be implemented in a number of different Structures or realizations. The choice of the structure can influence the computational complexity (amount of arithmetic operations per iteration) of the process and also the necessary number of iterations to achieve a desired performance level. Basically, there are two major classes of adaptive digital filter realizations, distinguished by the form of the impulse response, namely the finite-duration impulse response (FIR) filter and the infinite-duration impulse response (IIR) filters. FIR filters are usually implemented with non recursive structures, whereas IIR filters utilize recursive realizations.

Least Mean Square Adaptive Finite Impulse Response Filters

Basic Wiener Filter Theory In signal processing, the Wiener filter is a filter proposed by Norbert Wiener during the 1940s and published in 1949. Its purpose is to reduce the amount of noise present in a signal by comparison with an estimation of the desired noiseless signal. The discrete-time equivalent of Wiener's work was derived independently by Kolmogorov and published in 1941. Hence the

theory is often called the Wiener Kolmogorov filtering theory. The Wiener Kolmogorov was the first statistically designed filter to be proposed and subsequently gave rise to many others including the famous Kalman filter. A Wiener filter is not an adaptive filter because the theory behind this filter assumes that the inputs are stationary. Description The goal of the Wiener filter is to filter out noise that has corrupted a signal. It is based on a statistical approach. Typical filters are designed for a desired frequency response. However, the design of the Wiener filter takes a different approach. One is assumed to have knowledge of the spectral properties of the original signal and the noise, and one seeks the linear time-invariant filter whose output would come as close to the original signal as possible. Wiener filters are characterized by the following. Assumption: signal and (additive) noise are stationary linear stochastic processes with known spectral characteristics or known autocorrelation and cross-correlation Requirement: the filter must be physically realizable/causal (this requirement can be dropped, resulting in a non-causal solution) Performance criterion: minimum mean-square error (MMSE) This filter is frequently used in the process of deconvolution for this application, see Wiener deconvolution. Wiener filter problem setup The input to the Wiener filter is assumed to be a signal, , corrupted by additive noise, The output, , is calculated by means of a filter, , using the following convolution .

where

is the original signal (not exactly known; to be estimated) is the noise is the estimated signal (the intention is to equal )

is the Wiener filter's impulse response

The error is defined as

where

is the delay of the Wiener filter (since it is causal)

In other words, the error is the difference between the estimated signal and the true signal shifted by . The squared error is

where

is the desired output of the filter is the error , the problem can be described as follows:

Depending on the value of

If then the problem is that of prediction (error is reduced when is similar to a later value of s) If then the problem is that of filtering (error is reduced when is similar to ) If then the problem is that of smoothing (error is reduced when is similar to an earlier value of s) as a convolution integral:

Writing

Taking the expected value of the squared error results in

where

is the observed signal

is the autocorrelation function of is the autocorrelation function of is the cross-correlation function of

and is zero), then

If the signal and the noise this means that

are uncorrelated (i.e., the cross-correlation

For many applications, the assumption of uncorrelated signal and noise is reasonable. The goal is to minimize , the expected value of the squared error, by finding the optimal , the Wiener filter impulse response function. The minimum may be found by calculating the first order incremental change in the least square error resulting from an incremental change in g(.) for positive time. This is

For a minimum, this must vanish identically for all

which leads to the Wiener-Hopf equation

This is the fundamental equation of the Wiener theory. The right-hand side resembles a convolution but is only over the semi-infinite range. The equation can be solved by a special technique due to Wiener and Hopf

Implementation of adaptive filter through matlab

Application of adaptive filters a) Noise Cancellation The use of adaptive filters for reducing the noise content is based on the assumption that the frequency content of the event shall be unique from the background noise. This is readily justified for the case in which the background noise is continuous and the event is transient. The transient behavior implies that the frequency content of the event shall be spread out over many frequency bins due to its impulsive temporal characteristics. Additionally, for many sources of background noise, the spectral content is quite low. For engine based noise, the signal is inherently periodic in nature based on the primary excitation modes of the rotating structures. The frequency content between engine types and configurations can vary a great deal. The spectral signature of a single cylinder engine on a test stand is considerably simpler than a turbine or 16 cylinder diesels with their associated gear trains. The frequency components for these sources can be isolated and matched to appropriately tuned FIR or IIR filter banks to reduce their amplitude. The feed forward variety (FIR) formulations offer a higher degree of stability but typically require many more taps to realize a given frequency response. The basic formulation of an LMS adaptive FIR filter is

Where

This iterative adaptation of the weights utilizes gradient descent to assign filter tap coefficients such that the observed difference between the filter and the desired output are minimized. There

are two key tuning parameters that directly affect the convergence behavior; the number of filter taps used and the step size m . This algorithm is typically extended to include a normalization factor to the weight adaptation:

Additionally, a leakage factor may be introduced to allow the filter to "forget" its learned weights over time. This helps the weights adapt to different learning modes and avoid local minima.

. There are several variations on the filter tap update algorithms, including the Recursive Least Square, Fast Newton Transversal Filter, Affine Projection Algorithm, and the Fast Affine Projection. All are viable candidates for noise reduction with varying convergence and computation characteristics. To evaluate the applicability of adaptive filters to suppression of background noise for transient event identification, a dataset of background noises and events were obtained. The background noises were of 4 main classes; boats, diesel engines, jet engines, and helicopters. For each of these four classes 3 distinct examples were obtained, bringing the number of background noises to 12. The transient events that were obtained were of two footsteps, one lock being turned, one glass breaking, one door creaking, and one hammer striking. The event amplitudes were then reduced and combined with the various background noises to define signatures which were dominated by the noise. The SNR of these corrupted events ranged from -43dB to -22dB.

, where x is the event, M = number of samples in event, y is the noise and N = 1sec worth of samples (22000).

A total of 72 noise corrupted signatures were processed using adaptive filters to determine the applicability of adaptive filters to this problem domain. This dataset was processed using an adaptive FIR containing 91 taps, m = .5 with a = 1.0. b) Echo Cancellation in Telephonic Circuits The echo canceller's goal is to detect and remove echo as quickly and effectively, thereby minimizing any loss in voice quality due to the echo. The echo canceller must perform this function under all conditions including double-talk (when both parties are speaking at the same time) and in the presence of background noise. Furthermore, the echo canceller must not cause detriment to signaling tones (DTMF, etc.) or fax and modem transmissions. Telephone Circuit - Phone A's transmission passes through Hybrid A, through Echo Canceller A, through the Telephone Network, through Echo Canceller B, through Hybrid B to telephone B. A similar path is established in between Phone B and Phone A. When Phone A's transmission reaches Hybrid B, part of the signal is reflected by the hybrid back towards Echo Canceller B and therefore back to Phone A. If echo cancellation is not performed (and the network delay is moderate), the speaker at Phone A will perceive his echo. It is the responsibility of Echo Canceller B to cancel the echo that is induced by Hybrid B. Likewise, it is the responsibility of Echo Canceller A to cancel the echo that is induced by Hybrid A. the terminology Near End and Far End are usually used when referring to an echo canceller. For example, the Far End signal enters echo canceller A and passes through unchanged and is sent out to the hybrid. The hybrid, which is at the Near End with respect to echo canceller A, reflects a portion of the far end signal back towards the echo canceller. The Near End signal received by echo canceller A therefore consists of the sum of Phone A's transmit signal and the echo of the far end induced by Hybrid at different levels Adaptive Filtering As each hybrid circuit is slightly different each echo tail is different as well. Many factors determine the echo path. It is even possible for an echo tail to change while a circuit is active. This could happen when a second telephone extension is taken off-hook in parallel with the first one. Due to these variations in echo tails, it is necessary for an echo canceller to adapt to the tail continuously. Adaptive Filtering is employed within echo cancellers

to this end. The adaptive filters should converge quickly, but not so quickly that they might diverge under some conditions. This is especially important when a circuit is first established. The amount of time it takes the echo canceller to adapt to an echo path is referred to as the "convergence time".