Finite Impulse Response (FIR)

Filter
Dr. Dur-e-Shahwar Kundi
Lec-7
 Filter Design Steps
The design of digital filter involves five steps:
• Specification of filter requirements.
• Calculation of suitable filter coefficients.
• Realization of filter by a suitable structure.
• Analysis of effects of finite word length on filter performance
• Implementation of filter in software or hardware
 Specification of Filter Requirement
• The amplitude-frequency response of FIR/IIR filter is specified in
the form of tolerance scheme.

• Figure depicts the tolerance

scheme for lowpass filter.
– δp : Peak passband deviation
– δs : Peak stopband deviation
– fp : Passband edge frequency
– fs : Stopband edge frequency

• The width of transition band

determines how sharp the filter is.
Specification of Filter Requirement
• Figure depicts the tolerance scheme for bandpass filter. Following
parameters are of interest:
– δp : Peak passband deviation (ripples)
– δs : Peak stopband deviation
– fp1 & fp2 : Passband edge frequencies
– fs1 & fs2 : Stopband edge frequencies
• The width of transition band determines how sharp the filter is

4
Example:
Solution:
Coefficient Calculation
• Select one of approximation methods to calculate
coefficient values:
– h(k) for FIR
– ak and bk for IIR

• Methods to calculate FIR coefficients:

– Window method (simple & flexible)
– Frequency Sampling (recursive realization but lacks flexibility)
– Optimal (efficient & easy to use)

• Methods to calculate IIR coefficients:

– Impulse Invariant
– Bilinear Transformation
– pole-zero placement
 Filter Design
Frequency
Response
Magnitude/Phase
Analysis of
Designed Filter
Stability, Filter
Realizations
Filter
IIR (Bilinear
Method)
Designing of New
Filter
FIR (Window
Method)
 Analysis of Designed Filter
Digital Filter Frequency Response:
• Using the Laplace transfer function, we can
achieve the analog filter steady-state
frequency response H( jω) by substituting s=
jω into the transfer function H(s). i.e.

• Then |H( jω)| gives the magnitude frequency

response and H( jω) the phase response of
analog filter.
• Similarly, in a DSP system, we substitute
into z-transfer function H(z) to
acquire the digital frequency response.

• Then |H(ejωT)| will gives the magnitude

frequency response and |H(ejωT)| phase
response of digital filter.
• Let us introduce normalized digital frequency
in radians in digital domain as follows

• Then the digital frequency response will be

represented as
Example 6.10: (Li Tan Pg:183)
• Given the following difference equation of Filter system with
a sampling rate of 8,000 Hz, Determine the frequency
response of the given FIR filter.
y(n)=0.5x(n) + 0.5x(n-1)

1. First convert the given filter equation into H(z)

2. Then substitute
• Then magnitude and phase response will be

• Calculate several point as follows

• Plot of frequency response
Example 6.11: (Li Tan Pg:186)
• Given the following difference equation of
Filter system with a sampling rate of 8,000
Hz, Determine the frequency response of the
given IIR filter.
y(n)=x(n) - 0.5y(n-1)
 Designing of FIR Filter
window method
• We start from the ideal frequency response of the
filter i.e. HD(ω).
• Taking the Laplace inverse, we will get its ideal
impulse response i.e. hD(n) as shown
Summary of Ideal Impulse Response
for FIR Filter
• If we observe hD(n), it decreases if we move away
from n=0 and continuous to n=±ꝏ, thus resulting
filter is not FIR.
• Ideal solution is to truncate the hD(n)=0 for n>M, i.e.
multiplying it with suitable window function w(n)
having finite duration.
• Therefore final desired filter h(n) will be obtained as
follows

h(n)= hD(n) × w(n)

Summary of common window
function
Further Kaiser Window
Parameters:
Example:
Solution:
• Step1: Identify the ideal impulse response of desired Filter i.e.
Lowpass

• Step2: Based on the stopband attenuation (A) select an

appropriate window function w(n) i.e. Hamming, Blackman,
Kaiser satisfy the A > 50dB.
• Step3: Finding the value of “N” based on the normalized
transition width (∆f) i.e.
∆f =3.3/N (from widow function)
0.5KHz/8KHz =3.3/N
N= 52.8 = 53

• Step4: As the impulse response of the filter is symmetric

about the origin so the number of impulses “n” ranges
-[(N-1)/2] ≤ n ≤ +[(N-1)/2]
-26 ≤ n ≤ +26

• Step5: Defining the cutoff Frequency for h(n) i.e. fc’

fc’ = fc + ∆f/2
= 1.5KHz + (0.5KHz/2)
= 1.75 KHz / 8 KHz (normalized)
= 0.21875
• Step6: As we know that h(n) is symmetric so we need to
compute the impulse response for positive values of “n” i.e. n
= 0, 1,2, …… 26 and use symmetry property to obtain other
coefficients h(-1)=h(1), h(-2)=h(2) and so on
• Step7: As we note that indices of filter run from -26 to 26. To
make filter causal we add 26 to each index so that it starts
from zero. The resulting filter coefficient with indices adjusted
are
• Final filter equation:

h(n) = b0 x(n)+ b1 x(n-1) + b2 x(n-2) +………+ b26 x(n-26)+……..+

b51 x(n-51)+ b52 x(n-52)

H(z) = b0 z-0 + b1 z-1 + b2 z-2 +………+ b26 z-26 +……..+ b51 z-51 +
b52 z-52
Example:
 Fourier Transform Method
• An ideal lowpass filter with a normalized cutoff frequency Ωc,
whose magnitude frequency response in terms of the
normalized digital frequency Ω is characterized and plotted.

• The frequency response is periodic with a period of Ω=2π

radians
• The periodic frequency response can be
approximated using a complex Fourier series
expansion in terms of the normalized digital
frequency Ω

• Fourier coefficients are given by

• The fundamental frequency

• Substituting ω0 = 1 and cn = h(n) called the desired
impulse response of the ideal filter, we obtain the
Fourier transform design as

• If we substitute ejΩ = z and ω0 = 1 into Equation of it

will yield a z-transfer function
• The desired impulse response approximation of the
ideal lowpass filter is solved as
• The FIR filter design must first be completed by
truncating the infinite-length sequence h(n) to
achieve the 2M+1 dominant coefficients using the
coefficient symmetry

• For Causal FIR filter

• Delay operation is given by

Summary of Ideal Impulse Response
for FIR Filter