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Published byNoah Parsons Modified over 9 years ago
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Finite Impuse Response Filters
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Filters A filter is a system that processes a signal in some desired fashion. –A continuous-time signal or continuous signal of x(t) is a function of the continuous variable t. A continuous-time signal is often called an analog signal. –A discrete-time signal or discrete signal x(kT) is defined only at discrete instances t=kT, where k is an integer and T is the uniform spacing or period between samples
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Types of Filters There are two broad categories of filters: –An analog filter processes continuous-time signals –A digital filter processes discrete-time signals. The analog or digital filters can be subdivided into four categories: –Lowpass Filters –Highpass Filters –Bandstop Filters –Bandpass Filters
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Ideal Filters PassbandStopband Passband Stopband Lowpass FilterHighpass Filter Bandstop Filter PassbandStopband Bandpass Filter M( ) c c c 1 c 1 c 2 c 2
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Discrete-Time Signals Discrete-Time System T{ } x[n] y[n]=T{x[n]} inputoutput The diagram suggests that the output sequence is related to the input sequence by a process that can be described mathematically by an operator T.
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Moving Average Filter A simple, but useful filter is the moving average filter. Assume we have the following inputs, x[n]: 2 4 6
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A 3-point average for a finite-length signal of the values {x[0], x[1], x[2]} = {2, 4, 6} gives the answer ⅓ (2+4+6) = 4. This value defines one of the output values. The next output value is obtained by averaging {x[1], x[2], x[3]} = {4, 6, 4} that yields a value of 14/3.
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y[0] = ⅓(x[0] + x[1] + x[2]) y[1] = ⅓ (x[1] + x[2] + x[3]) which generalizes to the following input-output equation y[n] = ⅓ (x[n] + x[n+1] + x[n+2]) This equation is called a difference equation.
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For the triangular input, the result is the signal y[n] as tabulated below: Note that the values bold type in the x[n] row are the numbers involved in the computation of y[2].
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The output sequence is plotted below: 2 4 6 Note that the output sequence is longer than the input sequence and somewhat rounded.
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In general, values from either the present or future or both can be used FIR filter calculations. A filter that uses only the present and past values of the input is called a causal filter. A filter that uses future values of the input is called a non-causal filter. An alternative output indexing scheme can produce a filter that is causal.
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For the previous problem, the output becomes: Note that this form is simply a time-shifted form of the original form.
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The General FIR Filter The general form for the FIR filter is:
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Implementation of FIR Filters Recall that the general form for the FIR filter is:
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The Unit Impulse The unit impulse is perhaps the simplest sequence because it has only one non-zero value, which occurs at t = 0. The mathematical notation is: 1
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The Shifted Impulse 1 k The shifted impulse, [n-k], is non-zero when its argument is zero, i.e., n-k = 0, or when n = 0.
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The shifted impulse is a very useful concept for representing signals and systems. For example, 2 4 6
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In order to implement this form, we need the following: (1) a means of multiplying delayed-input signals by the filter coefficients; (2) a means of adding the scaled sequence values; (3) a means of obtaining delayed versions of the input sequence.
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It is useful to represent these operations in block diagram form. × x[n]y[n] y[n] = x[n] + Multiplier x 1 [n] x 2 [n] y[n] Adder y[n] = x 1 [n] + x 2 [n] x[n]y[n] Delay y[n] = x[n-1] Unit Delay
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Example: The FIR filter is completely defined once the set of filter coefficients {b k } is known. For example, if the {b k } are then we have a length 4 filter with M = 3. This expands into a 4- point difference equation:
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To illustrate the utility of the results that we obtained, consider the cascade of two systems defined by: The overall cascade system has the impulse response convolution
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In the analog world, convolution is described by the equation: Constant with respect to . Rotated about the y-axis. Moves along the x-axis.
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In order to find the overall impulse response we must convolve h 1 [n] with h 2 [n]. Thus, the equivalent impulse response is n0 1 2 3 4 5 h_1[n]1 1 1 1 h_2[n]1 1 1 h_1[0] h_2[n]1 1 1 1 h_1[1] h_2[n] 1 1 1 1 h_1[2] h_2[n] 1 1 1 1 h[n]1 2 3 3 2 1
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Thus, the equivalent impulse response is where {b k } is the sequence {1, 2, 3, 3, 2, 1}. This means that a system with impulse response h[n] can be implemented by the single difference equation
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Use convolution to compute the output y[n] for the length 4 filter that have the coefficients b k = {1, -2, 2, -1}. Use the input signal shown below.
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Find the system function H(z) of a FIR filter whose impulse response is:
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Compuational structure for a first order FIR filter. (a) The equivalence between z -1 and the unit delay; (b) Block diagram for the first order filter whose difference equation is y[n] = b 0 x[n] + b 1 x[n-1]. Draw a block diagram similar to (b) for the first difference system: y[n] = (1- z -1 ){x[n]}.
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