NH 67, Karur Trichy Highways, Puliyur C.F, Karur District DEPARTMENT OF INFORMATION TECHNOLOGY DIGITAL SIGNAL PROCESSING UNIT 3

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1 NH 67, Karur Trichy Highways, Puliyur C.F, Karur District DEPARTMENT OF INFORMATION TECHNOLOGY DIGITAL SIGNAL PROCESSING UNIT 3 IIR FILTER DESIGN Structure of IIR System design of Discrete time IIR filter from continuous time filter IIR filter design by Impulse invariance. Bilinear transformation Approximation derivatives Design of IIR filter in the Frequency domain. STRUCTURE OF IIR Infinite impulse response (IIR) is a property of signal processing systems. Systems with that property are known as IIR systems or when dealing with electronic filter systems as IIR filters. They have an impulse response function which is non-zero over an infinite length of time. This is in contrast to finite impulse response filters (FIR) which have fixed-duration impulse responses. The simplest analog IIR filter is an RC filter made up of a single resistor (R) feeding into a node shared with a single capacitor (C). This filter has an exponential impulse response characterized by an RC time constant. IIR filters may be implemented as either analog or digital filters. In digital IIR filters, the output feedback is immediately apparent in the equations defining the output. Note that unlike with FIR filters, in designing IIR filters it is necessary to carefully consider "time zero" case in which the outputs of the filter have not yet been clearly defined. Design of digital IIR filters is heavily dependent on that of their analog counterparts because there are plenty of resources, works and straightforward design methods concerning analog feedback filter design while there are hardly any for digital IIR filters. As a result, mostly, if a digital IIR filter is going to be

2 implemented, first, an analog filter (e.g. Chebyshev filter, Butterworth filter, Elliptic filter) is designed and then it is converted to digital by applying discretization techniques such as Bilinear transform or Impulse invariance. In practice, electrical engineers find IIR filters to be fast and cheap, but with poorer bandpass filtering and stability characteristics than FIR filters. Example IIR filters include the Chebyshev filter, Butterworth filter, and the Bessel filter. In the following subsections we focus on discrete time IIR filters which can be implemented in Digital Signal Processors. The frequency response of the Butterworth filter is maximally flat (has no ripples) in the passband, and rolls off towards zero in the stopband. When viewed on a logarithmic Bode plot, the response slopes off linearly towards negative infinity. For a first-order filter, the response rolls off at 6 db per octave ( 20 db per decade) (all first-order filters, regardless of name, have the same normalized frequency response). For a second-order Butterworth filter, the response decreases at 12 db per octave, a third-order at 18 db, and so on. Butterworth filters have a monotonically changing magnitude function with ω. The Butterworth is the only filter that maintains this same shape for higher orders (but with a steeper decline in the stopband) whereas other varieties of filters (Bessel, Chebyshev, elliptic) have different shapes at higher orders. Compared with a Chebyshev Type I/Type II filter or an elliptic filter, the Butterworth filter has a slower roll-off, and thus will require a higher order to implement a particular stopband specification. However, Butterworth filter will have a more linear phase response in the passband than the Chebyshev Type I/Type II and elliptic filters.

3 IIR vs FIR Filters 2

4 Difference equation: IIR as a class of LTI Filters Transfer function: To give an Infinite Impulse Response (IIR), a filter must be recursive, that is, incorporate feedback N 0, M 0 the recursive (previous output) terms feed back energy into the filter input and keep it going. (Although recursive filters are not necessarily IIR) 3

5 IIR Filters Design from an Analogue Prototype Given filter specifications, direct determination of filter coefficients is too complex Well-developed design methods exist for analogue low-pass filters Almost all methods rely on converting an analogue filter to a digital one 4

6 Analogue filter Rational Transfer Function ( ) 5

7 Analogue to Digital Conversion Im (z) Re (z) 6

8 Impulse Invariant method 7

9 Impulse Invariant method: Steps 1. Compute the Inverse Laplace transform to get impulse response of the analogue filter 2. Sample the impulse response (quickly enough to avoid aliasing problem) 3. Compute z-transform of resulting sequence 8

10 Example 1 Impulse Invariant Method Consider first order analogue filter = 1 - Corresponding impulse response is δ - v The presence of delta term prevents sampling of impulse response which thus cannot be defined Fundamental problem: high-pass and band-stop filters have functions with numerator and denominator polynomials of the same degree and thus cannot be designed using this method 9

11 Example 2 Impulse Invariant Method Consider an analogue filter Step 1. Impulse response of the analogue filter Step 2. Sample the impulse response Step 3. Compute z- transform The poles are mapped as 10

12 Impulse Invariant Method Indeed, in the general case the poles are mapped as k since any rational transfer function with the numerator degree strictly less than the denominator degree can be decomposed to partial fractions k k and similarly it can be shown k k 11

13 Impulse Invariant Method: Stability Since poles are mapped as: k k stable analogue filter is transformed into stable digital filter k < 1 12

14 Summary of the Impulse Invariant Method 13

15 Summary of the Impulse Invariant Method Advantage: preserves the order and stability of the analogue filter Disadvantages: Not applicable to all filter types (high-pass, band-stop) There is distortion of the shape of frequency response due to aliasing not in common use 14

16 Matched z-transform method 15

17 Example of Impulse Invariant vs Matched z transform methods 16

18 Backward Difference Method The analogue-domain variable s represents differentiation. We can try to replace s by approximating differentiation operator in the digital domain: Thus, y(t) = y(n) Y(z) X(z) Which suggests the s-to-z transformation: delay backward difference operator 17

19 Backward Difference Operator 18

20 Backward Difference method - Stability = = 0.5 The Left half s-plane onto the interior of the circle with radius 0.5 and centre at 0.5 in the z- plane Stable analogue filters become stable digital filters. However, poles are conned to a relatively small set of frequencies, no highpass filter possible! 19

21 Summary of the Backward Difference method Since the imaginary axis in the s domain are not mapped to the unit circle we can expect that the frequency response will be considerably distorted An analogue high-pass filter cannot be mapped to a digital high-pass because the poles of the digital filter cannot lie in the correct region method is crude and rarely used 20

22 Bilinear transform Bilinear transform is a correction of the backwards difference method The bilinear transform (also known as Tustin's transformation) is defined as the substitution: It is the most popular method The bilinear transform produces a digital filter whose frequency response has the same characteristics as the frequency response of the analogue filter (but its impulse response may then be quite different). 21

23 The bilinear transform The bilinear transform Note 1. Although the ratio could have been written (z-1)/(z+1), that causes unnecessary algebra later, when converting the resulting transfer function into a digital filter; Note 2. In some sources you will see the factor (2/T) multiplying the RHS of the bilinear transform; this is an optional scaling, but it cancels and does not affect the final result. 22

24 Where is the Bilinear Transform coming from? 23

25 24 To derive the properties of the bilinear transform, solve for z, and put ( ) ( ) hence ; ω ω ω ω = + + = + = a a z j a j a s s z Properties of the Bilinear Transform

26 Properties of the Bilinear Transform Look at two important cases: 1. The imaginary axis, i.e. =0. This corresponds to the boundary of stability for the analogue filter s poles. With =0, we have the imaginary (frequency) axis in the s-plane maps to the unit circle in the z-plane 2. With <0, i.e. the left half-plane in the s-plane we have left half s-plane maps onto the interior of the unit circle 25

27 Properties of the Bilinear Transform Thus the bilinear transform maps the Left half s-plane onto the interior of the unit circle in the z-plane: s-plane z-plane 1 1 This property allows us to obtain a suitable frequency response for the digital filter, and also to ensure the stability of the digital filter. 26

28 Properties of the Bilinear Transform z -1 s = z +1 = = ( ) 27

Properties of the Bilinear Transform Hence the Bilinear Transform preserves the following important features of the frequency response: 1. the Ω ω mapping is monotonic, and 2.

29 Properties of the Bilinear Transform Hence the Bilinear Transform preserves the following important features of the frequency response: 1. the Ω ω mapping is monotonic, and 2. Ω= 0 is mapped to ω = 0, and Ω = is mapped to ω = π (half the sampling frequency). Thus, for example, a low-pass response that decays to zero at Ω = produces a low-pass digital filter response that decays to zero at ω = π. 3. Mapping between the frequency variables is 28

30 Properties of the Bilinear Transform If the frequency response of the analogue filter at frequency Ω is H(jΩ), then the frequency response of the digital filter at the corresponding frequency ω = 2 arctan(ω) is also H(j Ω). Hence -3dB frequencies become -3dB frequencies, minimax responses remain minimax, etc. 29

31 Proof of Stability of the FIlter Suppose the analogue prototype H(s) has a stable pole at, i.e. Then the digital filter is obtained by substituting Since H(s) has a pole at, has a pole at because However, we know that lies within the unit circle. Hence the filter is guaranteed stable provided H(s) is stable. 30

32 Frequency Response of the Filter The frequency response of the analogue filter is The frequency response of the digital filter is Hence we can see that the frequency response is warped by a function Analogue Frequency Digital Frequency 31 31

33 Design using the bilinear transform The steps of the bilinear transform method are as follows: 1. Warp the digital critical (e.g. band-edge or "corner") frequencies ω i, in other words compute the corresponding analogue critical frequencies Ω i = tan(ω i /2). 2. Design an analogue filter which satisfies the resulting filter response specification. 3. Apply the bilinear transform to the s-domain transfer function of the analogue filter to generate the required z-domain transfer function

34 Example: Application of Bilinear Transform Design a first order low-pass digital filter with -3dB frequency of 1kHz and a sampling frequency of 8kHz using a the first order analogue low-pass filter which has a gain of 1 (0dB) at zero frequency, and a gain of -3dB ( = 0.5 ) at Ω c rad/sec (the "cutoff frequency "). 33

35 Example: Application of Bilinear Transform First calculate the normalised digital cutoff frequency: 3dB cutoff frequency sampling frequency Calculate the equivalent pre-warped analogue filter cutoff frequency (rad/sec) Thus, the analogue filter has the system function 34

36 Example: Application of Bilinear Transform Apply Bilinear transform As a direct form implementation: Normalise to unity for recursive implementation Keep factorised to save one multiply 35

37 Example: Magnitude Frequency Response ω c = π/4 Note that the digital filter response at zero frequency equals 1, as for the analogue filter, and the digital filter response at ω = π equals 0, as for the analogue filter at Ω =. The 3dB frequency is ω c = π/4, as intended. 36

38 Example: Pole-zero diagram for digital design Note that: The filter is stable, as expected The design process has added an extra zero compared to the prototype - this is typical of filters designed by the bilinear transform. Imag(z) -1 O X 1 Re(z) 37

39 Example: Pole-zero diagram for digital design There is a Matlab routine BILINEAR which computes the bilinear transformation. The example above could be computed, for example, by typing [NUMd,DENd] = BILINEAR([0.4142],[ ],0.5) which returns NUMd = DENd =

40 Designing high-pass, band-pass and band-stop filters The previous examples we have discussed have concentrated on IIR filters with low-pass characteristics. There are various techniques available to transform a low-pass filter into a highpass/band-pass/band-stop filters. The most popular one uses a frequency transformation in the analogue domain. 39

41 Designing filters uding frequency transformation 40

42 Frequency transformations 41

43 Example. Frequency transformation 42

44 Example. Frequency transformation 43

45 Thank you! 44

LECTURER NOTE SMJE3163 DSP

LECTURER NOTE SMJE3163 DSP LECTURER NOTE SMJE363 DSP (04/05-) ------------------------------------------------------------------------- Week3 IIR Filter Design -------------------------------------------------------------------------