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Infinite Impulse Response (IIR) Filters
                 Vol-5
Introduction
   Infinite Impulse Response (IIR) filters are
    the first choice when:
        Speed is paramount.
        Phase non-linearity is acceptable.
   IIR filters are computationally more
    efficient than FIR filters as they require
    fewer coefficients due to the fact that they
    use feedback or poles.
   However feedback can result in the filter
    becoming unstable if the coefficients
    deviate from their true values.
Concept of Digital IIR filter

                    h(k), k=0,1,2,…,∞
     x(n)           (impulse response-             y(n)
Input sequence        infinite length)        output sequence




                 y(n)         h(k ) x(n k )
                        k 0
Properties of an IIR Filter
   The general equation of an IIR filter can
    be expressed as follows:
                    b0 b1 z 1  bN z N
              H z
                    1 a1 z 1  aM z M
                        N
                                   k
                            bk z
                        k 0
                          M
                                       k
                    1          ak z
                         k 1


   ak and bk are the filter coefficients.
Properties of an IIR Filter
   The transfer function can be factorised to
    give:
                z z1 z z 2  z z N   Y z
     H z   k
               z p1 z p2  z p N     X z


   Where: z1, z2, …, zN are the zeros,
           p1, p2, …, pN are the poles.
Properties of an IIR Filter .. continued

   Time domain theoretical equation of IIR

              y(n)           h(k ) x(n k )
                       k 0

   For the implementation of the above
    equation we need the difference equation:
               N                M
       y(n)       b x(n k )         b y (n k )
              k 0 k            k   1k
              b(0) x(n) b(1) x(n   1) b(2) x(n 2) 
                        a(1) y(n   1) a(2) y(n 2)
IIR difference Equation and Direct Implementation
                    Structure

           y(n) b(0) x(n) b(1) x(n 1) b(2) x(n 2) 
                          a(1) y(n 1) a(2) y(n 2)

   x(n)          b0                                   y(n)
                        +           +


          z-1                                   z-1


                 b1     +           +     a1


          z-1                                  z-1


                 b2                       a2



                IIR structure for order=N = M = 2
IIR Direct Form I and Direct Form II
              Structures
x[n]                                                     y[n]
                                +
                Z-1       b0                   Z-1

       x[n-1]                                          y[n-1]    IIR
                Z-1       b1
                                       -a1     Z-1              Direct
       x[n-2]                                                   Form I
                                                     y[n-2]
                          b2
                                       -a2


       x[n]           +                       +          y[n]
                                      b0
                                Z-1                           IIR Direct
                          -a1         b1                       Form II
                                Z-1
                                                              (Canonic)
                          -a2         b2


  y(n) b(0)x(n) b(1)x(n 1) b(2)x(n 2)  a(1) y(n 1) a(2) y(n 2)
Design Procedure
   To fully design and implement a filter five
    steps are required:
     (1)   Filter specification.
     (2)   Coefficient calculation.
     (3)   Structure selection.
     (4)   Simulation (optional).
     (5)   Implementation.
Filter Specification - Step 1
  |H(f )|     pass-band                        stop-band

      1




                            fc : cut-of f f requency               fs/2       f (norm)
                                        (a)


  |H(f )|     pass-band   transition band        stop-band                |H(f )|
   (dB)                                                                   (linear)
          p                                                               1       p

      0                                                                   1
                                                                          1           p

                                                       pass-band
     -3
                                                         ripple



                                                       stop-band
                                                         ripple
          s                                                                   s



                                                                   fs/2       f (norm)
                                   fsb : stop-band f requency
                                fc : cut-of f f requency
                              fpb : pass-band f requency
                                        (b)
Coefficient Calculation - Step 2
   There are two different methods available
    for calculating the coefficients:
        Direct placement of poles and zeros.
        Using analogue filter design.
            Impulse invariant
            Bilinear z-transform etc
Pole-zero Placement Method
   All that is required for this method is the
    knowledge that:
        Placing a zero near or on the unit circle in
         the z-plane will minimise the transfer
         function at this point.
        Placing a pole near or on the unit circle in
         the z-plane will maximise the transfer
         function at this point.
        To obtain real coefficients the poles and
         zeros must either be real or occur in
         complex conjugate pairs.
Analogue to Digital Filter Conversion
   This is one of the simplest method.
   There is a rich collection of prototype
    analogue filters with well-established
    analysis methods.
   The method involves designing an
    analogue filter and then transforming it to
    a digital filter.
   The two principle methods are:
        Bilinear transform method
        Impulse invariant method.
Realisation Structures - Step 3
   Direct Form I:
                         N
                                        k
                                 bk z
          Y z            k 0                    b0 b1 z 1  bN z N
    H z                    M
          X z                               k   1 a1 z 1  aM z M
                     1            ak z
                          k 1


   Difference equation:
                             N                    M
                yn                bk x n k              ak y n k
                          k 0                     k 1




   This leads to the following structure…
Realisation Structures - Step 3
   Direct Form I:
       x(n)                                     y(n)
                    b0     +   +


              z-1                         z-1


                    b1     +   +   a1


              z-1                         z-1


                    b2     +   +   a2




                    bN-1   +   +   aM-1


              z-1                         z-1


                    bN     +   +   aM
Realisation Structures - Step 3
   Direct Form II canonic realisation:
    x(n)                    P(n)              y(n)
           +                         b0   +

                      z-1
                            P(n-1)
           +    -a1                  b1   +

                      z-1
                            P(n-2)
           +    -a2                  b2   +




                      z-1
                            P(n-N)
           +   -aN                   bN   +
Bilinear Transform Method
   Practical example of the bilinear
    transform method:
        The design of a digital filter to approximate a
         second order low-pass analogue filter is
         required.
        The transfer function that describes the
         analogue filter is (This is an analog
         BUTTERWORTH filter):
                               1
                    H s
                          s2    2s 1
        The digital filter is required to have:
            Cut-off frequency =100 Hz
            Sampling frequency of 1 kHz.
Example: Design of a LP filter design using
        Bilinear Transformation

  Design a digital equivalent of a 2nd order
  Butterworth LP filter with a
  cut-off freq fc=100 Hz
  Sampling frequency fs=1000 Hz.
  The normalized cut off frequency of the digital filter
  ω=2πfc/fs=2πfc 100/1000=0.628
  Now equivalent analog filter cut-off freq
  Ω=ktan(ω/2)= 1.tan(0.628/2)=0.325 rads/sec
Design of a LP filter design using Bilinear
         Transformation ..continued
     H(s) for a Butterworth filter
                            1
             H ( s)
                      s2      2s 1
Now the transfer function of this Butterworth filter
becomes (putting s=s/Ω)
                                 1
             H ( s)
                        s 2              s
                                     2         1
                      0.325            0.325
Design of a LP filter design using Bilinear Transformation
                         ..continued
 Now convert the analog filter H(s) into equivalent
 digital filter H(z) by applying the bilinear z-transform-
                                 1
                       z 1 1 z
                   s             1
                       z 1 1 z

                             1
      H ( z)
                  1   1 z 1 2    2 1 z 1
                                              1
               0.3252 1 z 1   0.325 1 z 1

               0.067 0.135z 1 0.067 z 2
               1 1.1429 z 1 0.4127 z 2
Design of a LP filter design using Bilinear Transformation
                         ..continued
     From the previous we get the filter coefficients-
    {bk}Coefficients          {ak}coefficients
    b0=0.067                  a1= -1.1429
    b1=0.0135                 a2=0.4127
    b2=0.067

  The time domain difference equation is obtained
    y(n) b(0) x(n) b(1) x(n 1) b(2) x(n 2) 
                    a(1) y(n 1) a(2) y(n 2)
         0.67 x(n) 0.135x(n 1) 0.67 x(n)
                   1.1429 y(n 1) 0.4127 y(n 2)
Direct realization of the design

x(n)                                                y(n)
               b0       +   +

             b0=0.067
       z-1                                    z-1


                b       +   +        a
                 1                    1
             b1=0.135           a1=1.1429
       z-1                                    z-1


                b                    a
                 2                    2
             b2=0.067           a2= -0.4127
Matlab Code for the BZT Method
clear all
fc=100;fs=1000;
wp=2*pi*fc*(1/fs);
OmegaP=2*fs*tan(wp/2);
Omega=tan(wp/2);
Hs_den1=[(1/Omega)^2,((2^0.5)/Omega),1];
Hs_den=[(1/OmegaP)^2,((2^0.5)/OmegaP),1];
Hs_num=1;
Hs_denN=Hs_den/Hs_den(1);
Hs_numN=Hs_num/Hs_den(1);
[b,a]=bilinear(Hs_numN,Hs_denN,fs);
freqz(b,a,fs)
Infinite Impulse Response (IIR) Filters
                - End -

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Dsp lecture vol 5 design of iir

  • 1. Infinite Impulse Response (IIR) Filters Vol-5
  • 2. Introduction  Infinite Impulse Response (IIR) filters are the first choice when:  Speed is paramount.  Phase non-linearity is acceptable.  IIR filters are computationally more efficient than FIR filters as they require fewer coefficients due to the fact that they use feedback or poles.  However feedback can result in the filter becoming unstable if the coefficients deviate from their true values.
  • 3. Concept of Digital IIR filter h(k), k=0,1,2,…,∞ x(n) (impulse response- y(n) Input sequence infinite length) output sequence y(n) h(k ) x(n k ) k 0
  • 4. Properties of an IIR Filter  The general equation of an IIR filter can be expressed as follows: b0 b1 z 1  bN z N H z 1 a1 z 1  aM z M N k bk z k 0 M k 1 ak z k 1  ak and bk are the filter coefficients.
  • 5. Properties of an IIR Filter  The transfer function can be factorised to give: z z1 z z 2  z z N Y z H z k z p1 z p2  z p N X z  Where: z1, z2, …, zN are the zeros, p1, p2, …, pN are the poles.
  • 6. Properties of an IIR Filter .. continued  Time domain theoretical equation of IIR y(n) h(k ) x(n k ) k 0  For the implementation of the above equation we need the difference equation: N M y(n) b x(n k ) b y (n k ) k 0 k k 1k b(0) x(n) b(1) x(n 1) b(2) x(n 2)  a(1) y(n 1) a(2) y(n 2)
  • 7. IIR difference Equation and Direct Implementation Structure y(n) b(0) x(n) b(1) x(n 1) b(2) x(n 2)  a(1) y(n 1) a(2) y(n 2) x(n) b0 y(n) + + z-1 z-1 b1 + + a1 z-1 z-1 b2 a2 IIR structure for order=N = M = 2
  • 8. IIR Direct Form I and Direct Form II Structures x[n] y[n] + Z-1 b0 Z-1 x[n-1] y[n-1] IIR Z-1 b1 -a1 Z-1 Direct x[n-2] Form I y[n-2] b2 -a2 x[n] + + y[n] b0 Z-1 IIR Direct -a1 b1 Form II Z-1 (Canonic) -a2 b2 y(n) b(0)x(n) b(1)x(n 1) b(2)x(n 2)  a(1) y(n 1) a(2) y(n 2)
  • 9.
  • 10. Design Procedure  To fully design and implement a filter five steps are required: (1) Filter specification. (2) Coefficient calculation. (3) Structure selection. (4) Simulation (optional). (5) Implementation.
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  • 12.
  • 13. Filter Specification - Step 1 |H(f )| pass-band stop-band 1 fc : cut-of f f requency fs/2 f (norm) (a) |H(f )| pass-band transition band stop-band |H(f )| (dB) (linear) p 1 p 0 1 1 p pass-band -3 ripple stop-band ripple s s fs/2 f (norm) fsb : stop-band f requency fc : cut-of f f requency fpb : pass-band f requency (b)
  • 14. Coefficient Calculation - Step 2  There are two different methods available for calculating the coefficients:  Direct placement of poles and zeros.  Using analogue filter design.  Impulse invariant  Bilinear z-transform etc
  • 15. Pole-zero Placement Method  All that is required for this method is the knowledge that:  Placing a zero near or on the unit circle in the z-plane will minimise the transfer function at this point.  Placing a pole near or on the unit circle in the z-plane will maximise the transfer function at this point.  To obtain real coefficients the poles and zeros must either be real or occur in complex conjugate pairs.
  • 16. Analogue to Digital Filter Conversion  This is one of the simplest method.  There is a rich collection of prototype analogue filters with well-established analysis methods.  The method involves designing an analogue filter and then transforming it to a digital filter.  The two principle methods are:  Bilinear transform method  Impulse invariant method.
  • 17. Realisation Structures - Step 3  Direct Form I: N k bk z Y z k 0 b0 b1 z 1  bN z N H z M X z k 1 a1 z 1  aM z M 1 ak z k 1  Difference equation: N M yn bk x n k ak y n k k 0 k 1  This leads to the following structure…
  • 18. Realisation Structures - Step 3  Direct Form I: x(n) y(n) b0 + + z-1 z-1 b1 + + a1 z-1 z-1 b2 + + a2 bN-1 + + aM-1 z-1 z-1 bN + + aM
  • 19. Realisation Structures - Step 3  Direct Form II canonic realisation: x(n) P(n) y(n) + b0 + z-1 P(n-1) + -a1 b1 + z-1 P(n-2) + -a2 b2 + z-1 P(n-N) + -aN bN +
  • 20. Bilinear Transform Method  Practical example of the bilinear transform method:  The design of a digital filter to approximate a second order low-pass analogue filter is required.  The transfer function that describes the analogue filter is (This is an analog BUTTERWORTH filter): 1 H s s2 2s 1  The digital filter is required to have:  Cut-off frequency =100 Hz  Sampling frequency of 1 kHz.
  • 21. Example: Design of a LP filter design using Bilinear Transformation Design a digital equivalent of a 2nd order Butterworth LP filter with a cut-off freq fc=100 Hz Sampling frequency fs=1000 Hz. The normalized cut off frequency of the digital filter ω=2πfc/fs=2πfc 100/1000=0.628 Now equivalent analog filter cut-off freq Ω=ktan(ω/2)= 1.tan(0.628/2)=0.325 rads/sec
  • 22. Design of a LP filter design using Bilinear Transformation ..continued H(s) for a Butterworth filter 1 H ( s) s2 2s 1 Now the transfer function of this Butterworth filter becomes (putting s=s/Ω) 1 H ( s) s 2 s 2 1 0.325 0.325
  • 23. Design of a LP filter design using Bilinear Transformation ..continued Now convert the analog filter H(s) into equivalent digital filter H(z) by applying the bilinear z-transform- 1 z 1 1 z s 1 z 1 1 z 1 H ( z) 1 1 z 1 2 2 1 z 1 1 0.3252 1 z 1 0.325 1 z 1 0.067 0.135z 1 0.067 z 2 1 1.1429 z 1 0.4127 z 2
  • 24. Design of a LP filter design using Bilinear Transformation ..continued From the previous we get the filter coefficients- {bk}Coefficients {ak}coefficients b0=0.067 a1= -1.1429 b1=0.0135 a2=0.4127 b2=0.067 The time domain difference equation is obtained y(n) b(0) x(n) b(1) x(n 1) b(2) x(n 2)  a(1) y(n 1) a(2) y(n 2) 0.67 x(n) 0.135x(n 1) 0.67 x(n) 1.1429 y(n 1) 0.4127 y(n 2)
  • 25. Direct realization of the design x(n) y(n) b0 + + b0=0.067 z-1 z-1 b + + a 1 1 b1=0.135 a1=1.1429 z-1 z-1 b a 2 2 b2=0.067 a2= -0.4127
  • 26. Matlab Code for the BZT Method clear all fc=100;fs=1000; wp=2*pi*fc*(1/fs); OmegaP=2*fs*tan(wp/2); Omega=tan(wp/2); Hs_den1=[(1/Omega)^2,((2^0.5)/Omega),1]; Hs_den=[(1/OmegaP)^2,((2^0.5)/OmegaP),1]; Hs_num=1; Hs_denN=Hs_den/Hs_den(1); Hs_numN=Hs_num/Hs_den(1); [b,a]=bilinear(Hs_numN,Hs_denN,fs); freqz(b,a,fs)
  • 27. Infinite Impulse Response (IIR) Filters - End -