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DSP - Digital Signal ProcessingDSP - Digital Signal Processing
Part 3Part 3
Dr. Krishnanaik VankdothDr. Krishnanaik Vankdoth
B.EB.E(ECE),(ECE), M.TechM.Tech (ECE),(ECE), Ph.DPh.D (ECE)(ECE)
Professor in ECE Dept
Aksum University, Ethiopia– 1010
Dr. V. Krishnanaik Ph.D
Books.Books.
1.1. Digital Signal Processing Principles, Algorithms and ApplicationsDigital Signal Processing Principles, Algorithms and Applications
John G.Proakis & Dimitris G.Manolakis
2.2. Digital Signal ProcessingDigital Signal Processing By.
Sen M. Kuo & Woon-Seng Gan
3.3. Digital Signal Processing A Practical Approach.Digital Signal Processing A Practical Approach. By
Emmanuel C. Ifeachor & Barrie W. Jervis
4.4. Digital Signal Processing By Dr. Krishnanaik Vankdoth LAPDigital Signal Processing By Dr. Krishnanaik Vankdoth LAP
LAMBERT Academic Publishing Dnfscland/Germany – 2014LAMBERT Academic Publishing Dnfscland/Germany – 2014
krishnanaik.ece@gmail.com 2
3
Grading Policy
krishnanaik.ece@gmail.com
Assignments 1 & 2
20 Marks
Quiz Test -Mid-Term 30 Marks
Final Exam 50 Marks
Class will be divided different level as per their GPA
Group A- GPA
Group B- GPA
Group C – GPA
4
DTFT and DFT
• The DTFT of an aperiodic discrete time
signal is defined as
• The DFT of a signal is defined as
• Inverse DFT is defined as
∑
∞
−∞=
−
=
n
jwn
enxwX ][][
∑
−
=
π−
=
1N
0n
N/n2jk
e]n[x)k(X
∑
−
=
=
1
0
/2
][
1
][
N
k
Nknj
ekX
N
nx π
What is difference between DTFT and DFT?
(1)
(2)
(3)
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5
• The DFT is periodic with period N.
Proof: ∑
−
=
π−
=
1N
0n
N/n2jk
e]n[x]k[X
∑
−
=
π+−
=+
1N
0n
N/n2)Nk(j
e]n[x]Nk[X
∑
−
=
π−π−
=
1N
0n
n2jN/n2jk
ee]n[x
Since e-j2πn
= 1
∑
−
=
π−
=+∴
1N
0n
N/n2jk
e]n[x]Nk[X
]k[X=
proved
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6
Example 1: Find the DFT of the following sequence
[1 0 0 1]
∑∑∑ =
π−
=
π−
−
=
π−
===
3
0n
2/njk
3
0n
4/n2jk
1N
0n
N/n2jk
e]n[xe]n[xe]n[x]k[X
21001]3[x]2[x]1[x]0[x]n[x]0[X
3
0i
=+++=+++==∑=
2/3j
3
0n
2/njk
e]3[x00]0[xe]n[x]1[X π−
=
π−
+++== ∑
j1)sin(j)cos(1e.11 2
3
2
32/3j
+=−+=+= πππ−
( ) 0]3sin)3.[cos(11]3[]0[][]2[
3
0
3
=−+=+== ∑=
−−
ππππ
jexxenxX
n
jnj
∑=
− −
− = + = =
3
0
2/ 9 2/ 3
1 ]3[ ]0[ ] [ ]3[
n
j n j
j e x x e n x Xπ π
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7
Example 2: Find the IDFT of the sequence
[2 1+j 0 1-j]
Solution:
Now
∑
−
=
π
=
1N
0k
N/n2jk
e]k[X
N
1
]n[x
[ ]]3[X]2[X]1[X]0[X
4
1
]k[X
4
1
]0[x
1N
0k
+++== ∑
−
=
∑∑ =
π
=
π
==
3
0k
2/jk
3
0k
4/2jk
e]k[X
4
1
e]k[X
4
1
]1[x
0e]3[Xe]2[Xe]1[X]0[X 2/3jj2/j
=+++= πππ
Similarly,
X[2] = 0 and X[3] = 1
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8
Computational Complexity of the DFT
A large number of multiplications and
additions are required for the calculation
of the DFT.
Consider an 8-point DFT as given by
Let k2π/8 = K
∑=
π−
=
7
0n
8/n2jk
e]n[x]k[X
∑=
−
=
7
0n
jKn
e]n[x]n[x
7jK6jK5jK
4jK3jK2jK1jK0jK
e]7[xe]6[xe]5[x
e]4[xe]3[xe]2[xe]1[xe]0[x
−−−
−−−−−
++
+++++
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9
There are eight complex multiplications and
seven complex additions. There are also
eight harmonic components to be evaluated.
Therefore, for an 8-point DFT:
Number of complex multiplications = 8×8
Number of complex additions = 8×7
For an N-point DFT
complex multiplications = N2
complex additions = N(N-1)
Clearly some means of reducing these is
required.
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10
Decimation-in-time fast fourier transform
algorithm (Cooley-Tuckey Algorithm):
Notations: Equation (2) can be re-written as
Let
N/nk2j
1n
0N
n1 ex]k[X π−
−
=
∑= (4)
N/2j
N eW π−
= (5)
Also note that 2/
)2//(22)/2(2
][ N
NjNj
N WeeW === −− ππ
(6)
and
)2/N)(N/2(jk
N
2/N
N
k
N
)2/Nk(
N eWWWW π−+
==
( ) k
N
k
N
jk
N WsinjcosWeW −=π−π== π−
(7)
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11
Summary:
N/2j
N eW π−
=
2/N
2
N WW =
k
N
)2/Nk(
N WW −=+
DFT: ∑
−
=
=
1N
0n
kn
Nn1 Wx]k[X (8)
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12
Consider n data samples as:
x0x1x2x3………xn
Divide these samples into an even
numbered and odd numbered sequenes x2n
and x2n+1 respectively.
That is,
x2n = x0x2x4…..,xN-2
x2n+1 = x1x3x5….xN-1
Both of the above sequences contain N/2
points. krishnanaik.ece@gmail.com
13
Now equation (8) can be re-written as
follows:
k)1n2(
N
12/N
0n
1n2
nk2
N
12/N
0n
n21 WxWx]k[X +
−
=
+
−
=
∑∑ +=
∑∑
−
=
+
−
=
+=
12/N
0n
nk2
N1n2
12/N
0n
k
N
nk2
n2 WxWWx N
since
nk
2/N
nk2
n WW =
Therefore, ∑∑
−
=
+
−
=
+=
12/N
0n
nk
2/N1n2
k
N
12/N
0n
nk
2/Nn21 WxWWx]k[X
The above equation can be re-written as
]k[XW]k[X]k[X 12
k
N111 += (9)
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14
Considering line 6 of the table it is seen that
4
k
4/N021 xwx]k[X += k = 0,1
Thus
4021 xx]0[X +=
while
404
2/2j
044/N021 xxxexxWx]1[X −=+=+= π−
similarly
7324
5123
6222
xx]0[X
xx]0[X
xx]0[X
+=
+=
+=
7324
5123
6222
xx]1[X
xx]1[X
xx]1[X
−=
−=
−=
We observe that the values with k = 1 differ only by a sign from
those with k = 0.
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15
Now ]k[XW]k[X]k[X 22
k
2/N2111 += (10)
So, ]0[X]0[X]0[XW]0[X]0[X 222122
0
2/N2111 +=+= (11)
]1[jX]1[Xe]1[X]1[XW]1[X]1[X 2221
2/j
2122
1
2/N2111 −=+=+= π−
(12)
]2[X]2[X]2[Xe]2[X]2[XW]2[X]2[X 222122
22)8/2(j
2122
2
2/N2111 −=+=+= ×π−
Now ]0[XxxxWxxWx]2[X 21404
2
204
2
2/N021 =+=+=+=
and ]0[XxxxWx]2[X 22626
2
4/N222 =+=+=
(13)
Hence equation (13) is equivalent to
]0[X]0[X]2[X 222111 +=
]3[XW]3[X]3[X 22
3
2/N2111 +=
(14)
(15)
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16
Now
]1[XxxxexxexxWx]3[X 21404
3j
04
3)2/2(j
04
3
4/N021 =−=+=+=+= π−π−
and ]1[Xxx]3[X 226222 =−=
Hence equation (15) is equivalent to
]1[jX]1[X]1[Xe]1[X]3[X 222122
3)4/2(j
2111 +=+= π−
(16)
Drawing these results together gives
]1[XW]1[X]1[jX]1[X]3[X
]1[XW]1[X]1[jX]1[X]1[X
]0[XW]0[X]0[X]0[X]2[X
]0[XW]0[X]0[X]0[X]0[X
22
2
821222111
22
2
821222111
22
0
821222111
22
0
821222111
−=+=
+=−=
−=−=
+=+=
(17)
The above equations are known as recomposition equations.
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17
The number of complex additions and
multiplications involved is reduced in this way
because:
(i) the recomposition equations are expressed in
terms of powers of the recurring factor WN.
(ii) use is also made of relationships of the type
X21[2] = X21[0] and X21[3] = X 21[1] and
(iii) the presence of only sign differences in the
pairs of expressions is exploited.
The algorithm is known as the Cooley-Tukey
algorithm.
It can be shown that
Number of complex multiplications = (N/2)log2N
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Frequency Domain Vs. Time Domain
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Fourier Transform
• A fourier transform is an useful analytical
tool that is important for many fields of
application in the digital signal processing.
• In describing the properties of the fourier
transform and inverse fourier transform, it
is quite convenient to use the concept of
time and frequency.
• In image processing applications it plays a
critical role. 19krishnanaik.ece@gmail.com
Fast Fourier transform
• Fast fourier transform proposed by Cooley
and Tukey in 1965.
• The fast fourier transform is a highly efficient
procedure for computing the DFT of a finite
series and requires less number of
computations than that of direct evaluation of
DFT.
• The FFT is based on decomposition and
breaking the transform into smaller transforms
and combining them to get the total transform.
20krishnanaik.ece@gmail.com
FFT
21krishnanaik.ece@gmail.com
DiscrDiscrete Fourier Transformete Fourier Transform
The DFT pair was given as
Baseline for computational complexity:
Each DFT coefficient requires
N complex multiplications
N-1 complex additions
All N DFT coefficients require
N2
complex multiplications
N(N-1) complex additions
[ ] ( )
∑
−
=
π
=
1N
0k
knN/2j
ekX
N
1
]n[x
[ ] ( )
∑
−
=
−
=
1
0
/2
][
N
n
knNj
enxkX π
22krishnanaik.ece@gmail.com
What is FFT?
• The fast fourier is an algorithm used to
compute the DFT. It makes use of the
symmetry and periodicity properties of twiddle
factor wN to effectively reduce the DFT
computation time.
• It is based on the fundamental principle of
decomposing the computation of DFT of a
sequence of length N into successively smaller
DFT.
23krishnanaik.ece@gmail.com
Symmetry and periodicity
Symmetry
Periodicit
y
)()(
)()(
)(
kNn
N
nNk
N
kn
N
nNk
N
Nnk
N
kn
N
kn
N
kn
N
WWW
WWW
WW
−−−
++
−∗
==
==
=
k
N
N/k
N
N/
N
mnk
mN
nk
N
mnk
mN
nk
N
WWW
WWWW
−=−=
==
+ )2(2
/
/
,1
,
24krishnanaik.ece@gmail.com
• FFT algorithm provides speed increase factors,
when compared with direct computation of the
DFT, of approximately 64 and 205 for 256
point and 1024 point transforms respectively.
• The number of multiplications and additions
required to compute N-point DFT using radix-
2 FFT are Nlog2N and N/2 log2N respectively.
25krishnanaik.ece@gmail.com
• Example:
The number of complex multiplications required
using direct computation is
N2
=642
=4096
The number of complex multiplications required
using FFT is
N/2log2N=64/2log264=192
Speed improvement factor =4096/192= 21.33.
26krishnanaik.ece@gmail.com
FFT Algorithms
• There are basically two types of FFT
algorithms.
• They are:
1. Decimation in Time
2. Decimation in frequency
27krishnanaik.ece@gmail.com
Decimation in time
• DIT algorithm is used to calculate the DFT of
a N-point sequence.
• The idea is to break the N-point sequence
into two sequences, the DFTs of which can
be obtained to give the DFT of the original
N-point sequence.
• Initially the N-point sequence is divided into
N/2-point sequences xe(n) and x0(n) ,
which have even and odd numbers of x(n)
respectively.
28krishnanaik.ece@gmail.com
• The N/2-point DFTs of these two sequences
are evaluated and combined to give the N-
point DFT.
• Similarly the N/2-point DFTs can be expressed
as a combination of N/4-point DFTs.
• This process is continued until we are left with
two point DFT.
• This algorithm is called decimation-in-time
because the sequence x(n) is often split into
smaller sequences.
29krishnanaik.ece@gmail.com
Radix-2 DIT- FFT Algorithm
Radix-2: the sequence length N
satisfied:
L is an integer
L
N 2=
 To decompose an N point time domain
signal into N signals each containing a
single point. Each decomposing stage
uses an interlace decomposition,
separating the even- and odd-indexed
samples;
 To calculate the N frequency spectra
corresponding to these N time domain
signals.
30krishnanaik.ece@gmail.com
Radix-2 DIT- FFT Algorithm
0 4 8 12 2 6 10 14 1 5 9 13 3 7 11 15
0 8 4 12 2 10 6 14 1 9 5 13 3 11 7 15
0 8 4 12 2 10 6 14 1 9 5 13 3 11 7 15
0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 151 signal of
16 points
0 2 4 6 8 10 12 14 1 3 5 7 9 11 13 152 signals
of 8 points
4 signals
of 4 points
8 signals
of 2 points
16 signals
of 1 point 31krishnanaik.ece@gmail.com
Radix-2 DIT- FFT Algorithm
 Algorithm principle
 To divide N-point sequence x(n) into two N/2-
point sequence x1(r) and x2(r)
1
2
,2,1,0,)12()();2()( 21 −=+==
N
rrxrxrxrx 
 To compute the DFT of x1(r) and x2(r)
)1
2
~0()12()()(
)1
2
~0()2()()(
1
2
0 2
1
2
0 2
22
1
2
0 2
1
2
0 2
11
−=+==
−===
∑∑
∑∑
−
=
−
=
−
=
−
=
N
kWrxWrxkX
N
kWrxWrxkX
N
r
rk
N
N
r
rk
N
N
r
rk
N
N
r
rk
N
32krishnanaik.ece@gmail.com
 To compute the DFT of N-point sequence x(n)
)1,2,1,0()()(
)()(
)12()2(
)()()()(
21
1
2
0 2
2
1
2
0 2
1
1
2
0
)12(
1
2
0
2
1
)(0
1
)(0
1
0
−=+=
+=
++=
+==
∑∑
∑∑
∑∑∑
−
=
−
=
−
=
+
−
=
−
=
−
=
−
=
NkkXWkX
WrxWWrx
WrxWrx
WnxWnxWnxkX
k
N
N
r
rk
N
k
N
N
r
rk
N
N
r
kr
N
N
r
rk
N
N
oddn
nk
N
N
evenn
nk
N
N
n
nk
N

33krishnanaik.ece@gmail.com
)1
2
,1,0()()(
)
2
()
2
()
2
(
)1
2
,1,0()()()(
21
2
)
2
(
1
21
−=−=
+++=+
−=+=
+
N
kkXWkX
N
kXW
N
kX
N
kX
N
kkXWkXkX
k
N
N
k
N
k
N


)(nx
)(1 rx
)(2 rx
)(1 kX
)(2 kX
)(kX
34krishnanaik.ece@gmail.com
 Butterfly computation flow graph
)1
2
,1,0()()()
2
(
)1
2
,1,0()()()(
21
21
−=−=+
−=+=
N
kkXWkX
N
kX
N
kkXWkXkX
k
N
k
N


)(1 kX
)(2 kX
k
NW
)()( 21 kXWkX k
N+
)()( 21 kXWkX k
N−
1−
There are 1 complex multiplication and 2 complex additions
35krishnanaik.ece@gmail.com
N/2-
point
DFT
N/2-
point
DFT
)0(1X
)1(1X
)2(1X
)3(1X
)0(2X
)1(2X
)2(2X
)3(2X
0
NW
1
NW
2
NW
3
NW
)0()0(1 xx =
)2()1(1 xx =
)4()2(1 xx =
)6()3(1 xx =
)1()0(2 xx =
)3()1(2 xx =
)5()2(2 xx =
)7()3(2 xx =
)(1 rx
)(2 rx
)4(X1−
)5(X1−
)6(X1−
)7(X1−
)0(X
)1(X
)2(X
)3(X
N-point DFT 36krishnanaik.ece@gmail.com
Radix-2 DIT- FFT Algorithm
fo
r
3
2=N
)(nx
2-point
DFT
2-point
DFT
2-point
DFT
2-point
DFT
Synthesize
the 2-point
DFTs into a
4-point DFT
Synthesize
the 2-point
DFTs into a
4-point DFT
Synthesize
the 4-point
DFTs into a
8-point DFT
)(kX
3-stage synthesize, each has N/2 butterfly
computation
 The computation complexity
37krishnanaik.ece@gmail.com
Radix-2 DIT- FFT Algorithm
•At the end of computation flow graph at any
stage, output variables can be stored in the
same registers previously occupied by the
corresponding input variables.
•This type of memory location sharing is
called in-place computation which results in
significant saving in overall memory
requirements.
38krishnanaik.ece@gmail.com
 The distance between two nodes in a butterfly
For there are L stagesL
N 2=
StageStage DistanceDistance
stage 1stage 1 11
stage 2stage 2 22
stage 3stage 3 44
stagestage LL

1
2 −L
39krishnanaik.ece@gmail.com
Radix-2 DIT- FFT Algorithm
 Bit-reversed order
In the DFT computation scheme, the DFT samples X(k)
appear at the output in a sequential order while the input
samples x(n) appear in a different order: a bit-reversed
order.
Thus, a sequentially ordered input x(n) must be
reordered appropriately before the fast algorithm can be
implemented.
Let m, n represent the sequential and bit-reversed order
in binary forms respectively, then:
m: 000 001 010 011 100 101 110 111
n: 000 100 010 110 001 101 011 111
40krishnanaik.ece@gmail.com
 Why is the input bit-reversed order
1n 2n
)( 012 nnnx
0n
0
1
0
1
0
1
0
1
0
1
0
1
0
1
)000(x
)100(x
)010(x
)110(x
)001(x
)101(x
)011(x
)111(x
)(0x
)(4x
)(2x
)(6x
)(1x
)(5x
)(3x
)(7x
41krishnanaik.ece@gmail.com
 How to get the bit-reversed order
Let represent the natural order, the represent
the bit-reversed order, then:
n nˆ
)ˆ()(ˆ nxnxnn ⇔> ,if
)0(A )1(A )2(A )3(A )4(A )5(A )6(A )7(A
)0(x )1(x )2(x )3(x )4(x )5(x )6(x )7(xn
nˆ )0(x )7(x)1(x)4(x )6(x)2(x )3(x)5(x
42krishnanaik.ece@gmail.com
Decimation-In-Frequency
• It is a popular form of FFT algorithm.
• In this the output sequence x(k) is divided into
smaller and smaller subsequences, that is why
the name decimation in frequency,
• Initially the input sequence x(n) is divided into
two sequences x1(n) and x2(n) consisting of
the first n/2 samples of x(n) and the last n/2
samples of x(n) respectively
43krishnanaik.ece@gmail.com
Radix-2 DIF- FFT Algorithm
 Algorithm principle
 To divide N-point sequence x(n) into two N/2-
point sequence
1
2
0),
2
(
1
2
0),(
−≤≤+
−≤≤
N
n
N
nx
N
nnxThe former N/2-point
The latter N/2-point
44krishnanaik.ece@gmail.com
0 1 2 3 4 5 6 7
0 1 2 3 4 5 6 7
0 1 2 3 0 1 2 3
butterfly computation
0 1 2 3 0 1 2 3
butterfly computation butterfly computation
0 1 0 1 0 1 0 1
butterfly butterfly butterfly butterfly
0 4 2 6 1 5 3 7
0 1 0 1 0 10 1
)(nx
)(kX 45krishnanaik.ece@gmail.com
 To compute the DFT of N-point sequence x(n)
)1,1,0()
2
()1()(
)
2
()(
)
2
()(
)()()()(
1
2
0
1
2
0
2
1
2
0
)
2
(
1
2
0
1
2
1
2
0
1
0
−=


 +−+=






++=
++=
+==
∑
∑
∑∑
∑∑∑
−
=
−
=
−
=
+
−
=
−
=
−
=
−
=
NkW
N
nxnx
W
N
nxWnx
W
N
nxWnx
WnxWnxWnxkX
N
n
nk
N
k
N
n
nk
N
k
N
N
N
n
k
N
n
N
N
n
nk
N
N
N
n
nk
N
N
n
nk
N
N
n
nk
N

46krishnanaik.ece@gmail.com
Radix-2 DIF- FFT Algorithm
 To separate the even and odd numbered
samples of X(k)
)1
2
,,1,0(,12,2let −=+==
N
rrkrk 
)1
2
,1,0()
2
()()2(
1
2
0 2
−=


 ++= ∑
−
=
N
rW
N
nxnxrX
N
n
nr
N 
)1
2
,1,0()
2
()()12(
1
2
0 2
−=


 +−=+ ∑
−
=
N
rWW
N
nxnxrX
N
n
nr
N
n
N 
47krishnanaik.ece@gmail.com
Radix-2 DIF- FFT Algorithm
)1
2
,1,0()()12(
)1
2
,1,0()()2(
1
2
0 2
2
1
2
0 2
1
−==+
−==
∑
∑
−
=
−
=
N
rWnxrX
N
rWnxrX
N
n
nr
N
N
n
nr
N


1
2
,1,0
)
2
()()(
)
2
()()(
let
2
1
−=












+−=
++=
N
n
W
N
nxnxnx
N
nxnxnx
n
N

48krishnanaik.ece@gmail.com
Radix-2 DIF- FFT Algorithm
)(nx
)
2
(
N
nx +
n
NW
1−
)
2
()()(1
N
nxnxnx ++=
n
NW
N
nxnxnx )
2
()()(2 





+−=
 Butterfly computation flow graph
There are 1 complex multiplication and 2 complex additions
49krishnanaik.ece@gmail.com
fo
r
3
2=N
N/2-
point
DFT
N/2-
point
DFT
)0(X
)2(X
)4(X
)6(X
)1(X
)3(X
)5(X
)7(X
)0(1x
)1(1x
)2(1x
)3(1x
)0(x
)1(x
)2(x
)3(x
)4(x
)5(x
)6(x
)7(x
)3(2x
3
NW
1−
)2(2x
2
NW
1−
)1(2x
1
NW
1−
)0(2x0
NW
1−
50krishnanaik.ece@gmail.com
fo
r
3
2=N
)0(x
)2(x
)1(x
)3(x
)4(x
)6(x
)5(x
)7(x
)0(X
)4(X
)2(X
)6(X
)1(X
)5(X
)3(X
)7(X
1−
1−
1−
1−
0
NW
1
NW
2
NW
3
NW
1−
1−
1−
1−
1−
0
NW
1−
0
NW
0
NW
2
NW
1−
0
NW
1−
0
NW
0
NW
2
NW
51krishnanaik.ece@gmail.com
Radix-2 DIF- FFT Algorithm
 The comparison of DIT and DIF
 The order of samples
DIT-FFT: the input is bit- reversed order and the
output is natural order
DIF-FFT: the input is natural order and the output is
bit- reversed order
 The butterfly computation
DIT-FFT: multiplication is done before additions
DIF-FFT: multiplication is done after additions
52krishnanaik.ece@gmail.com
Radix-2 DIF- FFT Algorithm
 Both DIT-FFT and DIF-FFT have the identical
computation complexity. i.e. for , there
are total L stages and each has N/2 butterfly
computation. Each butterfly computation has 1
multiplication and 2 additions.
L
N 2=
 Both DIT-FFT and DIF-FFT have the
characteristic of in-place computation.
 A DIT-FFT flow graph can be transposed to a
DIF-FFT flow graph and vice versa.
53krishnanaik.ece@gmail.com
krishnanaik.ece@gmail.com 54
krishnanaik.ece@gmail.com 55
Dr. V. Krishnanaik Ph.D
FIR and IIR Filter Design
Techniques
56
57
Outline
• Introduction
• IIR Filter Design by Impulse invariance
method
• IIR Filter Design by Bilinear transformation
method
• FIR Filter Design by Window function
technique
• FIR Filter Design by Frequency sampling
technique
• FIR Filter Design by MSEkrishnanaik.ece@gmail.com
krishnanaik.ece@gmail.com 58
Introduction
• Basic filter classification
• We put emphasis on the digital filter now,
and will introduce to the design method of
the FIR filter and IIR filter respectively.
Filter
Analog Filter
Digital Filter
IIR Filter
FIR Filter
59
• IIR is the infinite impulse response abbreviation.
• Digital filters by the accumulator, the multiplier,
and it constitutes IIR filter the way, generally
may divide into three kinds, respectively is
Direct form, Cascade form, and Parallel form.
• IIR filter design methods include the impulse
invariance, bilinear transformation, and step
invariance.
• We must emphasize at impulse invariance and
bilinear transformation.
krishnanaik.ece@gmail.com
krishnanaik.ece@gmail.com 60
Continuous frequency
band transformation
Impulse Invariance
method
Bilinear
transformation
method
Step invariance
method
IIR filter
Normalized analog
lowpass filter
IIR filter design methods
61
Introduction
• The structures of IIR filter
Direct
form 1
Direct form2
b0
b1
b2 b2
b1
b0
-a1
-a2
-a1
-a2
x(n) x(n)Y(n) Y(n)
1
z−
1
z−
1
z−
1
z−
1
z−
1
z−
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62
Introduction
• The structures of IIR filter
Cascade form
x(n) Y(n)
b0
b1
b2
-a1
-a2
-c1
-c2
d1
d2
Parallel form
Y(n)x(n)
b1
b0
d1
d0
E
-c1
-c2
-a1
-a2
1
z−
1
z−
1
z−
1
z−
1
z−
1
z−
1
z−
1
z−
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63
Introduction
• FIR is the finite impulse response
abbreviation, because its design
construction has not returned to the part
which gives.
• Its construction generally uses Direct form
and Cascade form.
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64
Introduction
• FIR filter design methods include the window
function, frequency sampling, minimize the
maximal error, and MSE.
• We must emphasize at window function,
frequency sampling, and MSE.
Window
function
technique
Frequency
sampling
technique
Minimize the
maximal error
FIR filter
Mean
square
error
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65
Introduction
• The structures of FIR filter
x(n) x(n)
b1
b2
b3
b4
b0
Y(n) Y(n)
Direct form Cascade form
b1
b2
d1
d2
b0
1
z−
1
z−
1
z−
1
z−
1
z−
1
z−
1
z−
1
z−
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66
IIR Filter Design by Impulse invariance
method
• The most straightforward of these is the
impulse invariance transformation
• Let be the impulse response
corresponding to , and define the
continuous to discrete time transformation by
setting
• We sample the continuous time impulse
response to produce the discrete time filter
( )ch t
( )cH s
( ) ( )ch n h nT=
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67
IIR Filter Design by Impulse invariance
method
• The frequency response is the Fourier
transform of the continuous time function
and hence
'( )H ω
*
( ) ( ) ( )c c
n
h t h nT t nTδ
∞
=−∞
= −∑
1 2
'( ) ( )c
k
H H j k
T T
π
ω ω
∞
=−∞
 
= − 
 
∑
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68
IIR Filter Design by Impulse invariance
method
• The system function is
• It is the many-to-one transformation from the
s plane to the z plane.
1 2
( ) | )sT cz e
k
H z H s jk
T T
π∞
=
=−∞
 
= − 
 
∑
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69
IIR Filter Design by Impulse invariance
method
• The impulse invariance transformation does
map the - axis and the left-half s plane
into the unit circle and its interior,
respectively
jω
Re(Z)
Im(Z)
1
S domain Z domain
sT
e
jω
σ
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70
IIR Filter Design by Impulse invariance
method
• is an aliased version of
• The stop-band characteristics are maintained adequately
in the discrete time frequency response only if the
aliased tails of are sufficiently small.
'( )H ω ( )cH jω
0 ω
'( )H ω
/Tπ 2 /Tπ
( )cH jω
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71
IIR Filter Design by Impulse invariance
method
• The Butterworth and Chebyshev-I lowpass
designs are more appropriate for impulse
invariant transformation than are the
Chebyshev-II and elliptic designs.
• This transformation cannot be applied
directly to highpass and bandstop designs.
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72
IIR Filter Design by Impulse invariance
method
• is expanded a partial fraction expansion
to produce
• We have assumed that there are no multiple
poles
• And thus
( )cH s
1
( )
N
k
c
k k
A
H s
s s=
=
−
∑
1
( ) ( )k
N
s t
c k
k
h t A e u t
=
= ∑
1
( ) ( )k
N
s nT
k
k
h n A e u n
=
= ∑
1
1
( )
1 k
N
k
s T
k
A
H z
e z−
=
=
−
∑
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73
IIR Filter Design by Impulse invariance
method
• Example:
Expanding in a partial fraction
expansion, it produce
The impulse invariant transformation
yields a discrete time design with the
system function
2 2
( )
( )
c
s a
H s
s a b
+
=
+ +
1/ 2 1/ 2
( )cH s
s a jb s a jb
= +
+ + + −
( ) 1 ( ) 1
1/ 2 1/ 2
( )
1 1a jb T a jb T
H z
e z e z− + − − − −
= +
− −
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74
IIR Filter Design by Bilinear
transformation method
• The most generally useful is the
bilinear transformation.
• To avoid aliasing of the frequency response as
encountered with the impulse invariance
transformation.
• We need a one-to-one mapping from the s plane
to the z plane.
• The problem with the transformation is
many-to-one.
sT
z e=
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75
IIR Filter Design by Bilinear
transformation method
• We could first use a one-to-one transformation
from to , which compresses the entire s
plane into the strip
• Then could be transformed to z by
with no effect from aliasing.
s 's
Im( ')s
T T
π π
− ≤ ≤
's
's T
z e=
σ
jω
'σ
jω
/Tπ−
/Tπ
s domain s’ domain
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76
IIR Filter Design by Bilinear
transformation method
• The transformation from to is given by
• The characteristic of this transformation is
seen most readily from its effect on the
axis.
• Substituting and , we obtain
s 's
12
' tanh ( )
2
sT
s
T
−
=
jω
s jω= ' 's jω=
12
' tan ( )
2
T
T
ω
ω −
=
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77
IIR Filter Design by Bilinear
transformation method
• The axis is compressed into the interval
for in a one-to-one method
• The relationship between and is
nonlinear, but it is approximately linear at
small .
( ,
T T
π π
−
'ω
ω
ω 'ω
'ω ω≈
-
ω
'ω
/Tπ
/Tπ−
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78
IIR Filter Design by Bilinear
transformation method
• The desired transformation to is now
obtained by inverting to produce
• And setting , which yields
12
' tanh ( )
2
sT
s
T
−
=
2 '
tanh( )
2
s T
s
T
=
s z
1
' ( )lns z
T
=
2 ln
tanh( )
2
z
s
T
=
1
1
2 1
( )
1
z
T z
−
−
−
=
+
Re(Z)
Im(Z)
1
S domain Z domain
1
2
1
2
T
s
z
T
s
+
=
−
jω
σ
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79
IIR Filter Design by Bilinear
transformation method
• The discrete-time filter design is obtained
from the continuous-time design by means of
the bilinear transformation
• Unlike the impulse invariant transformation,
the bilinear transformation is one-to-one, and
invertible.
1 1
(2/ )(1 )/(1 )
( ) ( ) |c s T z z
H z H s − −
= − +
=
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80
FIR Filter Design by Window function
technique
• Simplest FIR the filter design is window
function technique
• A supposition ideal frequency response may
express
where
( ) [ ]j j n
d d
n
H e h n eω ω
∞
−
=−∞
= ∑
1
[ ] ( )
2
j j n
d dh n H e e d
π
ω ω
π
ω
π −
= ∫
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81
FIR Filter Design by Window function
technique
• To get this kind of systematic causal FIR to
be approximate, the most direct method
intercepts its ideal impulse response!
[ ] [ ] [ ]dh n w n h n= g
( ) ( ) ( )dH W Hω ω ω= ∗
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82
FIR Filter Design by Window function
technique
• Truncation of the Fourier series produces
the familiar Gibbs phenomenon
• It will be manifested in , especially if
is discontinuous.
( )H ω
( )dH ω
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83
FIR Filter Design by Window function
technique
• 1.Rectangular window
• 2.Triangular window (Bartett window)
1, 0
[ ]
0,
n M
w n
otherwise
≤ ≤
= 

2 , 0
2
2[ ] 2 ,
2
0,
n Mn
M
n Mw n n M
M
otherwise
 ≤ ≤


= − < ≤


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84
FIR Filter Design by Window function
technique
• 1.Rectangular window
• 2.Triangular window (Bartett window)
0 10 20 30 40 50 60
0
0.5
1
sequence (n)
T(n)
Rectangular window
0 10 20 30 40 50 60
0
0.5
1
sequence (n)
T(n)
Bartlett window
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
-100
-50
0
50
100
pi unitsFrequencyresponseT(jw)(dB)
Rectangular window
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
-100
-50
0
50
100
pi units
FrequencyresponseT(jw)(dB)
Bartlett window
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85
FIR Filter Design by Window function
technique
• 3.HANN window
• 4.Hamming window
1 2
1 cos , 0
[ ] 2
0,
n
n M
w n M
otherwise
π  
− ≤ ≤  =  


2
0.54 0.46cos , 0
[ ]
0,
n
n M
w n M
otherwise
π
− ≤ ≤
= 

krishnanaik.ece@gmail.com
86
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
-100
-50
0
50
100
pi units
FrequencyresponseT(jw)(dB)
Hanning window
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
-100
-50
0
50
100
pi units
FrequencyresponseT(jw)(dB)
Hamming window
0 10 20 30 40 50 60
0
0.5
1
sequence (n)
T(n)
Hanning window
0 10 20 30 40 50 60
0
0.5
1
sequence (n)
T(n)
Hamming window
FIR Filter Design by Window function
technique
• 3.HANN window
• 4.Hamming window
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87
FIR Filter Design by Window function
technique
• 5.Kaiser’s window
• 6.Blackman window
2
0
0
2
[ 1 (1 ) ]
[ ] , 0,1,...,
[ ]
n
I
Mw n n M
I
β
β
− −
= =
2 4
0.42 0.5cos 0.08cos , 0
[ ]
0,
n n
n M
w n M M
otherwise
π π
− + ≤ ≤
= 

krishnanaik.ece@gmail.com
88
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
-100
-50
0
50
100
pi units
FrequencyresponseT(jw)(dB)
Blackman window
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
-150
-100
-50
0
50
100
pi units
FrequencyresponseT(jw)(dB)
Kaiser window
• 5.Kaiser’s window
• 6.Blackman window
0 10 20 30 40 50 60
0
0.5
1
sequence (n)
T(n)
Blackman window
0 10 20 30 40 50 60
0
0.5
1
sequence (n)
T(n)
Kaiser window
FIR Filter Design by Window function
technique
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89
FIR Filter Design by Window function
technique
( / )s Mω
Window Peak sidelobe level
(dB)
Transition
bandwidth
Max. stopband
ripple(dB)
Rectangular -13 0.9 -21
Hann -31 3.1 -44
Hamming -41 3.3 -53
Blackman -57 5.5 -74
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90
FIR Filter Design by Frequency sampling
technique
• For arbitrary, non-classical specifications of
, the calculation
of , n=0,1,…,M, via an appropriate
approximation can be a substantial
computation task.
• It may be preferable to employ a design
technique that utilizes specified values of
directly, without the necessity of
determining
' ( )dH ω
( )dh n
' ( )dH ω ( )dh n
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91
FIR Filter Design by Frequency sampling
technique
• We wish to derive a linear phase IIR filter
with real nonzero . The impulse
response must be symmetric
where are real and denotes the
integer part
( )h n
[ /2]
0
1
2 ( 1/ 2)
( ) 2 cos( )
1
M
k
k
k n
h n A A
M
π
=
+
= +
+
∑
kA [ / 2]M
0,1,...,n M=
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92
FIR Filter Design by Frequency sampling
technique
• It can be rewritten as
where and
• Therefore, it may write
where
1
/ 2 /
0
/2
( )
N
j k N j kn N
k
k
k N
h n A e eπ π
−
=
≠
= ∑ 0,1,..., 1n N= −
1N M= + k N kA A −=
/ 2 /
( ) j k N j kn N
k kh n A e eπ π
=
1
0
/2
( ) ( )
N
k
k
k N
h n h n
−
=
≠
= ∑
0,1,..., 1n N= −
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93
FIR Filter Design by Frequency sampling
technique
• with corresponding transform
where
• Hence
which has a linear phase
1
0
/2
( ) ( )
N
k
k
k N
H z H z
−
=
≠
= ∑
/
2 / 1
(1 )
( )
1
j k N N
k
k j k N
A e z
H z
e z
π
π
−
−
−
=
−
' ( 1)/2 sin / 2
( )
sin[( / / 2)]
j T N
k k
TN
H A e
k N T
ω ω
ω
π ω
− −
=
−
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94
FIR Filter Design by Frequency sampling
technique
• The magnitude response
which has a maximum value
at where
' sin / 2
( )
sin[( / / 2)]
k k
TN
H A
k N T
ω
ω
π ω
=
−
kN A
/k sk Nω ω= 2 /s Tω π=
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95
FIR Filter Design by Frequency sampling
technique
• The only nonzero contribution to at
is from , and hence that
• Therefore, by specifying the DFT samples of
the desired magnitude
response at the frequencies , and
setting
'( )H ω
kω ω= '
( )kH ω
'( )k kH N Aω =
'
( )dH ω kω
'
( ) /k d kA H Nω= ±
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96
FIR Filter Design by Frequency sampling
technique
• We produce a filter design from equation (5.1)
for which
• The desired and actual magnitude responses are
equal at the N frequencies
'
'( ) ( )k d kH Hω ω=
kω
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97
FIR Filter Design by Frequency sampling
technique
• In between these frequencies, is
interpolated as the sum of the responses
, and its magnitude does not, equal that of
'( )H ω
'
( )kH ω
'
( )dH ω
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98
FIR Filter Design by Frequency sampling
technique
• Example: For an ideal lowpass filter
from , we would choose
• The frequency samples are indeed equal
to the desired
' 1, 0,1,...,
( )
0, 1,...,[ / 2]
d k
k P
H
k P M
ω
=
= 
= +
'
( ) /k d kA H Nω= ±
( 1) / ( 1), 0,1,...,
0, 1,...,[ / 2]
k
k
M k P
A
k P M
 − + =
= 
= +
'
( )kH ω
'
( )d kH ω
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99
FIR Filter Design by Frequency sampling
technique
• The response is very similar to the result
form using the rectangular window, and the
stopband is similarly disappointing.
• We can try to search for the optimum value
of the transition sample would quickly lead
us to a value of approximately
, k p≠0.38( 1) /( 1)p
pA M= − +
krishnanaik.ece@gmail.com
100
FIR Filter Design by MSE
• : The spectrum of the filter we obtain
• : The spectrum of the desired filter
• MSE =
( )H f
( )dH f
( ) ( )∫−
−
−
2/
2/
21 s
s
f
f ds dffHfHf
0 0.1 0.2 0.3 0.4 0.5
-0.5
0
0.5
1
1.5
krishnanaik.ece@gmail.com
101
FIR Filter Design by MSE
• Larger MSE, but smaller maximal error
•
• Smaller MSE, but larger maximal error
0 0.1 0.2 0.3 0.4
-0.5
0
0.5
1
1.5
0 0.1 0.2 0.3 0.4
-0.5
0
0.5
H(F) H(F) - H (F)d
0 0.1 0.2 0.3 0.4
-0.5
0
0.5
1
1.5
0 0.1 0.2 0.3 0.4
-0.5
0
0.5
H(F) H(F) - H (F)d
krishnanaik.ece@gmail.com
102
FIR Filter Design by MSE
• 1.
( ) ( ) ( ) ( )∫∫ −−
−
−=−=
2/1
2/1
22/
2/
21
dFFHFRdffHfRfMSE d
f
f ds
s
s
( ) ( ) dFFHFnns d
k
n
∫ ∑−
=
−=
2/1
2/1
2
0
|| 2cos][ π
( ) ( ) ( ) ( ) dFFHFnnsFHFnns d
k
n
d
k
n
∫ ∑∑−
==






−





−=
2/1
2/1
00
2cos][2cos][ ππ
( ) ( )
1/2
1/2
0 0
[ ]cos 2 [ ]cos 2
k k
n
s n n F s F dF
τ
π τ π τ
−
= =
= ∑ ∑∫
( ) ( ) ( )
1/2 1/2
2
1/2 1/2
0
2 [ ]cos 2
k
d d
n
s n n F H F dF H F dFπ
− −
=
− +∑∫ ∫
krishnanaik.ece@gmail.com
103
FIR Filter Design by MSE
• 2. when n ≠ τ,
when n = τ, n ≠ 0,
when n = τ, n = 0,
• 3. The formula can be repressed as:
( ) ( ) 02cos2cos
2/1
2/1
=∫−
dFFFn τππ
( ) ( ) 2/12cos2cos
2/1
2/1
=∫−
dFFFn τππ
( ) ( ) 12cos2cos
2/1
2/1
=∫−
dFFFn τππ
( ) ( ) ( )dFFHdFFHFnnsnssMSE dd
k
n
k
n
∫∫ ∑∑ −−
==
+−+=
2/1
2/1
22/1
2/1
01
22
2cos][22/][]0[ π
krishnanaik.ece@gmail.com
104
FIR Filter Design by MSE
• 4. Doing the partial differentiation:
• 5. Minimize MSE: for all n’s
( )∫−
−=
∂
∂ 2/1
2/1
2]0[2
]0[
dFFHs
s
MSE
d ( ) ( )∫−
−=
∂
∂ 2/1
2/1
2cos2][
][
dFFHFnns
ns
MSE
dπ
0
][
=
∂
∂
ns
MSE
( )∫−
=
2/1
2/1
]0[ dFFHs d ( ) ( )∫−
=
2/1
2/1
2cos2][ dFFHFnns dπ
[ ] [0]
[ ] [ ]/ 2 for n=1,2,...,k
[ ] [ ]/ 2 for n=1,2,...,k
[ ] 0 for n<0 and n N
h k s
h k n s n
h k n s n
h n
=
+ =
− =
= ≥
krishnanaik.ece@gmail.com
105
• IIR advantage:
1. It is easy to design
2. It is easy to implementation
• IIR disadvantage:
1. Infinite impulse response
2. It is hard to optimalize than FIR
3. Non-stable
krishnanaik.ece@gmail.com
106Dr. V. Krishnanaik Ph.D
Dr. Krishnanaik VankdothDr. Krishnanaik Vankdoth
B.EB.E(ECE),(ECE), M.TechM.Tech (ECE),(ECE), Ph.DPh.D (ECE)(ECE)
Professor in ECE Dept
Aksum University, Ethiopia– 1010
Krishnanaik.ece@gmail.com
Krishnanaik_ece@yahoo.com
Phone : +919441629162
krishnanaik.ece@gmail.com 107

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Digital signal processor part 3

  • 1. 1 DSP - Digital Signal ProcessingDSP - Digital Signal Processing Part 3Part 3 Dr. Krishnanaik VankdothDr. Krishnanaik Vankdoth B.EB.E(ECE),(ECE), M.TechM.Tech (ECE),(ECE), Ph.DPh.D (ECE)(ECE) Professor in ECE Dept Aksum University, Ethiopia– 1010 Dr. V. Krishnanaik Ph.D
  • 2. Books.Books. 1.1. Digital Signal Processing Principles, Algorithms and ApplicationsDigital Signal Processing Principles, Algorithms and Applications John G.Proakis & Dimitris G.Manolakis 2.2. Digital Signal ProcessingDigital Signal Processing By. Sen M. Kuo & Woon-Seng Gan 3.3. Digital Signal Processing A Practical Approach.Digital Signal Processing A Practical Approach. By Emmanuel C. Ifeachor & Barrie W. Jervis 4.4. Digital Signal Processing By Dr. Krishnanaik Vankdoth LAPDigital Signal Processing By Dr. Krishnanaik Vankdoth LAP LAMBERT Academic Publishing Dnfscland/Germany – 2014LAMBERT Academic Publishing Dnfscland/Germany – 2014 krishnanaik.ece@gmail.com 2
  • 3. 3 Grading Policy krishnanaik.ece@gmail.com Assignments 1 & 2 20 Marks Quiz Test -Mid-Term 30 Marks Final Exam 50 Marks Class will be divided different level as per their GPA Group A- GPA Group B- GPA Group C – GPA
  • 4. 4 DTFT and DFT • The DTFT of an aperiodic discrete time signal is defined as • The DFT of a signal is defined as • Inverse DFT is defined as ∑ ∞ −∞= − = n jwn enxwX ][][ ∑ − = π− = 1N 0n N/n2jk e]n[x)k(X ∑ − = = 1 0 /2 ][ 1 ][ N k Nknj ekX N nx π What is difference between DTFT and DFT? (1) (2) (3) krishnanaik.ece@gmail.com
  • 5. 5 • The DFT is periodic with period N. Proof: ∑ − = π− = 1N 0n N/n2jk e]n[x]k[X ∑ − = π+− =+ 1N 0n N/n2)Nk(j e]n[x]Nk[X ∑ − = π−π− = 1N 0n n2jN/n2jk ee]n[x Since e-j2πn = 1 ∑ − = π− =+∴ 1N 0n N/n2jk e]n[x]Nk[X ]k[X= proved krishnanaik.ece@gmail.com
  • 6. 6 Example 1: Find the DFT of the following sequence [1 0 0 1] ∑∑∑ = π− = π− − = π− === 3 0n 2/njk 3 0n 4/n2jk 1N 0n N/n2jk e]n[xe]n[xe]n[x]k[X 21001]3[x]2[x]1[x]0[x]n[x]0[X 3 0i =+++=+++==∑= 2/3j 3 0n 2/njk e]3[x00]0[xe]n[x]1[X π− = π− +++== ∑ j1)sin(j)cos(1e.11 2 3 2 32/3j +=−+=+= πππ− ( ) 0]3sin)3.[cos(11]3[]0[][]2[ 3 0 3 =−+=+== ∑= −− ππππ jexxenxX n jnj ∑= − − − = + = = 3 0 2/ 9 2/ 3 1 ]3[ ]0[ ] [ ]3[ n j n j j e x x e n x Xπ π krishnanaik.ece@gmail.com
  • 7. 7 Example 2: Find the IDFT of the sequence [2 1+j 0 1-j] Solution: Now ∑ − = π = 1N 0k N/n2jk e]k[X N 1 ]n[x [ ]]3[X]2[X]1[X]0[X 4 1 ]k[X 4 1 ]0[x 1N 0k +++== ∑ − = ∑∑ = π = π == 3 0k 2/jk 3 0k 4/2jk e]k[X 4 1 e]k[X 4 1 ]1[x 0e]3[Xe]2[Xe]1[X]0[X 2/3jj2/j =+++= πππ Similarly, X[2] = 0 and X[3] = 1 krishnanaik.ece@gmail.com
  • 8. 8 Computational Complexity of the DFT A large number of multiplications and additions are required for the calculation of the DFT. Consider an 8-point DFT as given by Let k2π/8 = K ∑= π− = 7 0n 8/n2jk e]n[x]k[X ∑= − = 7 0n jKn e]n[x]n[x 7jK6jK5jK 4jK3jK2jK1jK0jK e]7[xe]6[xe]5[x e]4[xe]3[xe]2[xe]1[xe]0[x −−− −−−−− ++ +++++ krishnanaik.ece@gmail.com
  • 9. 9 There are eight complex multiplications and seven complex additions. There are also eight harmonic components to be evaluated. Therefore, for an 8-point DFT: Number of complex multiplications = 8×8 Number of complex additions = 8×7 For an N-point DFT complex multiplications = N2 complex additions = N(N-1) Clearly some means of reducing these is required. krishnanaik.ece@gmail.com
  • 10. 10 Decimation-in-time fast fourier transform algorithm (Cooley-Tuckey Algorithm): Notations: Equation (2) can be re-written as Let N/nk2j 1n 0N n1 ex]k[X π− − = ∑= (4) N/2j N eW π− = (5) Also note that 2/ )2//(22)/2(2 ][ N NjNj N WeeW === −− ππ (6) and )2/N)(N/2(jk N 2/N N k N )2/Nk( N eWWWW π−+ == ( ) k N k N jk N WsinjcosWeW −=π−π== π− (7) krishnanaik.ece@gmail.com
  • 11. 11 Summary: N/2j N eW π− = 2/N 2 N WW = k N )2/Nk( N WW −=+ DFT: ∑ − = = 1N 0n kn Nn1 Wx]k[X (8) krishnanaik.ece@gmail.com
  • 12. 12 Consider n data samples as: x0x1x2x3………xn Divide these samples into an even numbered and odd numbered sequenes x2n and x2n+1 respectively. That is, x2n = x0x2x4…..,xN-2 x2n+1 = x1x3x5….xN-1 Both of the above sequences contain N/2 points. krishnanaik.ece@gmail.com
  • 13. 13 Now equation (8) can be re-written as follows: k)1n2( N 12/N 0n 1n2 nk2 N 12/N 0n n21 WxWx]k[X + − = + − = ∑∑ += ∑∑ − = + − = += 12/N 0n nk2 N1n2 12/N 0n k N nk2 n2 WxWWx N since nk 2/N nk2 n WW = Therefore, ∑∑ − = + − = += 12/N 0n nk 2/N1n2 k N 12/N 0n nk 2/Nn21 WxWWx]k[X The above equation can be re-written as ]k[XW]k[X]k[X 12 k N111 += (9) krishnanaik.ece@gmail.com
  • 14. 14 Considering line 6 of the table it is seen that 4 k 4/N021 xwx]k[X += k = 0,1 Thus 4021 xx]0[X += while 404 2/2j 044/N021 xxxexxWx]1[X −=+=+= π− similarly 7324 5123 6222 xx]0[X xx]0[X xx]0[X += += += 7324 5123 6222 xx]1[X xx]1[X xx]1[X −= −= −= We observe that the values with k = 1 differ only by a sign from those with k = 0. krishnanaik.ece@gmail.com
  • 15. 15 Now ]k[XW]k[X]k[X 22 k 2/N2111 += (10) So, ]0[X]0[X]0[XW]0[X]0[X 222122 0 2/N2111 +=+= (11) ]1[jX]1[Xe]1[X]1[XW]1[X]1[X 2221 2/j 2122 1 2/N2111 −=+=+= π− (12) ]2[X]2[X]2[Xe]2[X]2[XW]2[X]2[X 222122 22)8/2(j 2122 2 2/N2111 −=+=+= ×π− Now ]0[XxxxWxxWx]2[X 21404 2 204 2 2/N021 =+=+=+= and ]0[XxxxWx]2[X 22626 2 4/N222 =+=+= (13) Hence equation (13) is equivalent to ]0[X]0[X]2[X 222111 += ]3[XW]3[X]3[X 22 3 2/N2111 += (14) (15) krishnanaik.ece@gmail.com
  • 16. 16 Now ]1[XxxxexxexxWx]3[X 21404 3j 04 3)2/2(j 04 3 4/N021 =−=+=+=+= π−π− and ]1[Xxx]3[X 226222 =−= Hence equation (15) is equivalent to ]1[jX]1[X]1[Xe]1[X]3[X 222122 3)4/2(j 2111 +=+= π− (16) Drawing these results together gives ]1[XW]1[X]1[jX]1[X]3[X ]1[XW]1[X]1[jX]1[X]1[X ]0[XW]0[X]0[X]0[X]2[X ]0[XW]0[X]0[X]0[X]0[X 22 2 821222111 22 2 821222111 22 0 821222111 22 0 821222111 −=+= +=−= −=−= +=+= (17) The above equations are known as recomposition equations. krishnanaik.ece@gmail.com
  • 17. 17 The number of complex additions and multiplications involved is reduced in this way because: (i) the recomposition equations are expressed in terms of powers of the recurring factor WN. (ii) use is also made of relationships of the type X21[2] = X21[0] and X21[3] = X 21[1] and (iii) the presence of only sign differences in the pairs of expressions is exploited. The algorithm is known as the Cooley-Tukey algorithm. It can be shown that Number of complex multiplications = (N/2)log2N krishnanaik.ece@gmail.com
  • 18. Frequency Domain Vs. Time Domain 18krishnanaik.ece@gmail.com
  • 19. Fourier Transform • A fourier transform is an useful analytical tool that is important for many fields of application in the digital signal processing. • In describing the properties of the fourier transform and inverse fourier transform, it is quite convenient to use the concept of time and frequency. • In image processing applications it plays a critical role. 19krishnanaik.ece@gmail.com
  • 20. Fast Fourier transform • Fast fourier transform proposed by Cooley and Tukey in 1965. • The fast fourier transform is a highly efficient procedure for computing the DFT of a finite series and requires less number of computations than that of direct evaluation of DFT. • The FFT is based on decomposition and breaking the transform into smaller transforms and combining them to get the total transform. 20krishnanaik.ece@gmail.com
  • 22. DiscrDiscrete Fourier Transformete Fourier Transform The DFT pair was given as Baseline for computational complexity: Each DFT coefficient requires N complex multiplications N-1 complex additions All N DFT coefficients require N2 complex multiplications N(N-1) complex additions [ ] ( ) ∑ − = π = 1N 0k knN/2j ekX N 1 ]n[x [ ] ( ) ∑ − = − = 1 0 /2 ][ N n knNj enxkX π 22krishnanaik.ece@gmail.com
  • 23. What is FFT? • The fast fourier is an algorithm used to compute the DFT. It makes use of the symmetry and periodicity properties of twiddle factor wN to effectively reduce the DFT computation time. • It is based on the fundamental principle of decomposing the computation of DFT of a sequence of length N into successively smaller DFT. 23krishnanaik.ece@gmail.com
  • 25. • FFT algorithm provides speed increase factors, when compared with direct computation of the DFT, of approximately 64 and 205 for 256 point and 1024 point transforms respectively. • The number of multiplications and additions required to compute N-point DFT using radix- 2 FFT are Nlog2N and N/2 log2N respectively. 25krishnanaik.ece@gmail.com
  • 26. • Example: The number of complex multiplications required using direct computation is N2 =642 =4096 The number of complex multiplications required using FFT is N/2log2N=64/2log264=192 Speed improvement factor =4096/192= 21.33. 26krishnanaik.ece@gmail.com
  • 27. FFT Algorithms • There are basically two types of FFT algorithms. • They are: 1. Decimation in Time 2. Decimation in frequency 27krishnanaik.ece@gmail.com
  • 28. Decimation in time • DIT algorithm is used to calculate the DFT of a N-point sequence. • The idea is to break the N-point sequence into two sequences, the DFTs of which can be obtained to give the DFT of the original N-point sequence. • Initially the N-point sequence is divided into N/2-point sequences xe(n) and x0(n) , which have even and odd numbers of x(n) respectively. 28krishnanaik.ece@gmail.com
  • 29. • The N/2-point DFTs of these two sequences are evaluated and combined to give the N- point DFT. • Similarly the N/2-point DFTs can be expressed as a combination of N/4-point DFTs. • This process is continued until we are left with two point DFT. • This algorithm is called decimation-in-time because the sequence x(n) is often split into smaller sequences. 29krishnanaik.ece@gmail.com
  • 30. Radix-2 DIT- FFT Algorithm Radix-2: the sequence length N satisfied: L is an integer L N 2=  To decompose an N point time domain signal into N signals each containing a single point. Each decomposing stage uses an interlace decomposition, separating the even- and odd-indexed samples;  To calculate the N frequency spectra corresponding to these N time domain signals. 30krishnanaik.ece@gmail.com
  • 31. Radix-2 DIT- FFT Algorithm 0 4 8 12 2 6 10 14 1 5 9 13 3 7 11 15 0 8 4 12 2 10 6 14 1 9 5 13 3 11 7 15 0 8 4 12 2 10 6 14 1 9 5 13 3 11 7 15 0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 151 signal of 16 points 0 2 4 6 8 10 12 14 1 3 5 7 9 11 13 152 signals of 8 points 4 signals of 4 points 8 signals of 2 points 16 signals of 1 point 31krishnanaik.ece@gmail.com
  • 32. Radix-2 DIT- FFT Algorithm  Algorithm principle  To divide N-point sequence x(n) into two N/2- point sequence x1(r) and x2(r) 1 2 ,2,1,0,)12()();2()( 21 −=+== N rrxrxrxrx   To compute the DFT of x1(r) and x2(r) )1 2 ~0()12()()( )1 2 ~0()2()()( 1 2 0 2 1 2 0 2 22 1 2 0 2 1 2 0 2 11 −=+== −=== ∑∑ ∑∑ − = − = − = − = N kWrxWrxkX N kWrxWrxkX N r rk N N r rk N N r rk N N r rk N 32krishnanaik.ece@gmail.com
  • 33.  To compute the DFT of N-point sequence x(n) )1,2,1,0()()( )()( )12()2( )()()()( 21 1 2 0 2 2 1 2 0 2 1 1 2 0 )12( 1 2 0 2 1 )(0 1 )(0 1 0 −=+= += ++= +== ∑∑ ∑∑ ∑∑∑ − = − = − = + − = − = − = − = NkkXWkX WrxWWrx WrxWrx WnxWnxWnxkX k N N r rk N k N N r rk N N r kr N N r rk N N oddn nk N N evenn nk N N n nk N  33krishnanaik.ece@gmail.com
  • 35.  Butterfly computation flow graph )1 2 ,1,0()()() 2 ( )1 2 ,1,0()()()( 21 21 −=−=+ −=+= N kkXWkX N kX N kkXWkXkX k N k N   )(1 kX )(2 kX k NW )()( 21 kXWkX k N+ )()( 21 kXWkX k N− 1− There are 1 complex multiplication and 2 complex additions 35krishnanaik.ece@gmail.com
  • 36. N/2- point DFT N/2- point DFT )0(1X )1(1X )2(1X )3(1X )0(2X )1(2X )2(2X )3(2X 0 NW 1 NW 2 NW 3 NW )0()0(1 xx = )2()1(1 xx = )4()2(1 xx = )6()3(1 xx = )1()0(2 xx = )3()1(2 xx = )5()2(2 xx = )7()3(2 xx = )(1 rx )(2 rx )4(X1− )5(X1− )6(X1− )7(X1− )0(X )1(X )2(X )3(X N-point DFT 36krishnanaik.ece@gmail.com
  • 37. Radix-2 DIT- FFT Algorithm fo r 3 2=N )(nx 2-point DFT 2-point DFT 2-point DFT 2-point DFT Synthesize the 2-point DFTs into a 4-point DFT Synthesize the 2-point DFTs into a 4-point DFT Synthesize the 4-point DFTs into a 8-point DFT )(kX 3-stage synthesize, each has N/2 butterfly computation  The computation complexity 37krishnanaik.ece@gmail.com
  • 38. Radix-2 DIT- FFT Algorithm •At the end of computation flow graph at any stage, output variables can be stored in the same registers previously occupied by the corresponding input variables. •This type of memory location sharing is called in-place computation which results in significant saving in overall memory requirements. 38krishnanaik.ece@gmail.com
  • 39.  The distance between two nodes in a butterfly For there are L stagesL N 2= StageStage DistanceDistance stage 1stage 1 11 stage 2stage 2 22 stage 3stage 3 44 stagestage LL  1 2 −L 39krishnanaik.ece@gmail.com
  • 40. Radix-2 DIT- FFT Algorithm  Bit-reversed order In the DFT computation scheme, the DFT samples X(k) appear at the output in a sequential order while the input samples x(n) appear in a different order: a bit-reversed order. Thus, a sequentially ordered input x(n) must be reordered appropriately before the fast algorithm can be implemented. Let m, n represent the sequential and bit-reversed order in binary forms respectively, then: m: 000 001 010 011 100 101 110 111 n: 000 100 010 110 001 101 011 111 40krishnanaik.ece@gmail.com
  • 41.  Why is the input bit-reversed order 1n 2n )( 012 nnnx 0n 0 1 0 1 0 1 0 1 0 1 0 1 0 1 )000(x )100(x )010(x )110(x )001(x )101(x )011(x )111(x )(0x )(4x )(2x )(6x )(1x )(5x )(3x )(7x 41krishnanaik.ece@gmail.com
  • 42.  How to get the bit-reversed order Let represent the natural order, the represent the bit-reversed order, then: n nˆ )ˆ()(ˆ nxnxnn ⇔> ,if )0(A )1(A )2(A )3(A )4(A )5(A )6(A )7(A )0(x )1(x )2(x )3(x )4(x )5(x )6(x )7(xn nˆ )0(x )7(x)1(x)4(x )6(x)2(x )3(x)5(x 42krishnanaik.ece@gmail.com
  • 43. Decimation-In-Frequency • It is a popular form of FFT algorithm. • In this the output sequence x(k) is divided into smaller and smaller subsequences, that is why the name decimation in frequency, • Initially the input sequence x(n) is divided into two sequences x1(n) and x2(n) consisting of the first n/2 samples of x(n) and the last n/2 samples of x(n) respectively 43krishnanaik.ece@gmail.com
  • 44. Radix-2 DIF- FFT Algorithm  Algorithm principle  To divide N-point sequence x(n) into two N/2- point sequence 1 2 0), 2 ( 1 2 0),( −≤≤+ −≤≤ N n N nx N nnxThe former N/2-point The latter N/2-point 44krishnanaik.ece@gmail.com
  • 45. 0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7 0 1 2 3 0 1 2 3 butterfly computation 0 1 2 3 0 1 2 3 butterfly computation butterfly computation 0 1 0 1 0 1 0 1 butterfly butterfly butterfly butterfly 0 4 2 6 1 5 3 7 0 1 0 1 0 10 1 )(nx )(kX 45krishnanaik.ece@gmail.com
  • 46.  To compute the DFT of N-point sequence x(n) )1,1,0() 2 ()1()( ) 2 ()( ) 2 ()( )()()()( 1 2 0 1 2 0 2 1 2 0 ) 2 ( 1 2 0 1 2 1 2 0 1 0 −=    +−+=       ++= ++= +== ∑ ∑ ∑∑ ∑∑∑ − = − = − = + − = − = − = − = NkW N nxnx W N nxWnx W N nxWnx WnxWnxWnxkX N n nk N k N n nk N k N N N n k N n N N n nk N N N n nk N N n nk N N n nk N  46krishnanaik.ece@gmail.com
  • 47. Radix-2 DIF- FFT Algorithm  To separate the even and odd numbered samples of X(k) )1 2 ,,1,0(,12,2let −=+== N rrkrk  )1 2 ,1,0() 2 ()()2( 1 2 0 2 −=    ++= ∑ − = N rW N nxnxrX N n nr N  )1 2 ,1,0() 2 ()()12( 1 2 0 2 −=    +−=+ ∑ − = N rWW N nxnxrX N n nr N n N  47krishnanaik.ece@gmail.com
  • 48. Radix-2 DIF- FFT Algorithm )1 2 ,1,0()()12( )1 2 ,1,0()()2( 1 2 0 2 2 1 2 0 2 1 −==+ −== ∑ ∑ − = − = N rWnxrX N rWnxrX N n nr N N n nr N   1 2 ,1,0 ) 2 ()()( ) 2 ()()( let 2 1 −=             +−= ++= N n W N nxnxnx N nxnxnx n N  48krishnanaik.ece@gmail.com
  • 49. Radix-2 DIF- FFT Algorithm )(nx ) 2 ( N nx + n NW 1− ) 2 ()()(1 N nxnxnx ++= n NW N nxnxnx ) 2 ()()(2       +−=  Butterfly computation flow graph There are 1 complex multiplication and 2 complex additions 49krishnanaik.ece@gmail.com
  • 52. Radix-2 DIF- FFT Algorithm  The comparison of DIT and DIF  The order of samples DIT-FFT: the input is bit- reversed order and the output is natural order DIF-FFT: the input is natural order and the output is bit- reversed order  The butterfly computation DIT-FFT: multiplication is done before additions DIF-FFT: multiplication is done after additions 52krishnanaik.ece@gmail.com
  • 53. Radix-2 DIF- FFT Algorithm  Both DIT-FFT and DIF-FFT have the identical computation complexity. i.e. for , there are total L stages and each has N/2 butterfly computation. Each butterfly computation has 1 multiplication and 2 additions. L N 2=  Both DIT-FFT and DIF-FFT have the characteristic of in-place computation.  A DIT-FFT flow graph can be transposed to a DIF-FFT flow graph and vice versa. 53krishnanaik.ece@gmail.com
  • 56. Dr. V. Krishnanaik Ph.D FIR and IIR Filter Design Techniques 56
  • 57. 57 Outline • Introduction • IIR Filter Design by Impulse invariance method • IIR Filter Design by Bilinear transformation method • FIR Filter Design by Window function technique • FIR Filter Design by Frequency sampling technique • FIR Filter Design by MSEkrishnanaik.ece@gmail.com
  • 58. krishnanaik.ece@gmail.com 58 Introduction • Basic filter classification • We put emphasis on the digital filter now, and will introduce to the design method of the FIR filter and IIR filter respectively. Filter Analog Filter Digital Filter IIR Filter FIR Filter
  • 59. 59 • IIR is the infinite impulse response abbreviation. • Digital filters by the accumulator, the multiplier, and it constitutes IIR filter the way, generally may divide into three kinds, respectively is Direct form, Cascade form, and Parallel form. • IIR filter design methods include the impulse invariance, bilinear transformation, and step invariance. • We must emphasize at impulse invariance and bilinear transformation. krishnanaik.ece@gmail.com
  • 60. krishnanaik.ece@gmail.com 60 Continuous frequency band transformation Impulse Invariance method Bilinear transformation method Step invariance method IIR filter Normalized analog lowpass filter IIR filter design methods
  • 61. 61 Introduction • The structures of IIR filter Direct form 1 Direct form2 b0 b1 b2 b2 b1 b0 -a1 -a2 -a1 -a2 x(n) x(n)Y(n) Y(n) 1 z− 1 z− 1 z− 1 z− 1 z− 1 z− krishnanaik.ece@gmail.com
  • 62. 62 Introduction • The structures of IIR filter Cascade form x(n) Y(n) b0 b1 b2 -a1 -a2 -c1 -c2 d1 d2 Parallel form Y(n)x(n) b1 b0 d1 d0 E -c1 -c2 -a1 -a2 1 z− 1 z− 1 z− 1 z− 1 z− 1 z− 1 z− 1 z− krishnanaik.ece@gmail.com
  • 63. 63 Introduction • FIR is the finite impulse response abbreviation, because its design construction has not returned to the part which gives. • Its construction generally uses Direct form and Cascade form. krishnanaik.ece@gmail.com
  • 64. 64 Introduction • FIR filter design methods include the window function, frequency sampling, minimize the maximal error, and MSE. • We must emphasize at window function, frequency sampling, and MSE. Window function technique Frequency sampling technique Minimize the maximal error FIR filter Mean square error krishnanaik.ece@gmail.com
  • 65. 65 Introduction • The structures of FIR filter x(n) x(n) b1 b2 b3 b4 b0 Y(n) Y(n) Direct form Cascade form b1 b2 d1 d2 b0 1 z− 1 z− 1 z− 1 z− 1 z− 1 z− 1 z− 1 z− krishnanaik.ece@gmail.com
  • 66. 66 IIR Filter Design by Impulse invariance method • The most straightforward of these is the impulse invariance transformation • Let be the impulse response corresponding to , and define the continuous to discrete time transformation by setting • We sample the continuous time impulse response to produce the discrete time filter ( )ch t ( )cH s ( ) ( )ch n h nT= krishnanaik.ece@gmail.com
  • 67. 67 IIR Filter Design by Impulse invariance method • The frequency response is the Fourier transform of the continuous time function and hence '( )H ω * ( ) ( ) ( )c c n h t h nT t nTδ ∞ =−∞ = −∑ 1 2 '( ) ( )c k H H j k T T π ω ω ∞ =−∞   = −    ∑ krishnanaik.ece@gmail.com
  • 68. 68 IIR Filter Design by Impulse invariance method • The system function is • It is the many-to-one transformation from the s plane to the z plane. 1 2 ( ) | )sT cz e k H z H s jk T T π∞ = =−∞   = −    ∑ krishnanaik.ece@gmail.com
  • 69. 69 IIR Filter Design by Impulse invariance method • The impulse invariance transformation does map the - axis and the left-half s plane into the unit circle and its interior, respectively jω Re(Z) Im(Z) 1 S domain Z domain sT e jω σ krishnanaik.ece@gmail.com
  • 70. 70 IIR Filter Design by Impulse invariance method • is an aliased version of • The stop-band characteristics are maintained adequately in the discrete time frequency response only if the aliased tails of are sufficiently small. '( )H ω ( )cH jω 0 ω '( )H ω /Tπ 2 /Tπ ( )cH jω krishnanaik.ece@gmail.com
  • 71. 71 IIR Filter Design by Impulse invariance method • The Butterworth and Chebyshev-I lowpass designs are more appropriate for impulse invariant transformation than are the Chebyshev-II and elliptic designs. • This transformation cannot be applied directly to highpass and bandstop designs. krishnanaik.ece@gmail.com
  • 72. 72 IIR Filter Design by Impulse invariance method • is expanded a partial fraction expansion to produce • We have assumed that there are no multiple poles • And thus ( )cH s 1 ( ) N k c k k A H s s s= = − ∑ 1 ( ) ( )k N s t c k k h t A e u t = = ∑ 1 ( ) ( )k N s nT k k h n A e u n = = ∑ 1 1 ( ) 1 k N k s T k A H z e z− = = − ∑ krishnanaik.ece@gmail.com
  • 73. 73 IIR Filter Design by Impulse invariance method • Example: Expanding in a partial fraction expansion, it produce The impulse invariant transformation yields a discrete time design with the system function 2 2 ( ) ( ) c s a H s s a b + = + + 1/ 2 1/ 2 ( )cH s s a jb s a jb = + + + + − ( ) 1 ( ) 1 1/ 2 1/ 2 ( ) 1 1a jb T a jb T H z e z e z− + − − − − = + − − krishnanaik.ece@gmail.com
  • 74. 74 IIR Filter Design by Bilinear transformation method • The most generally useful is the bilinear transformation. • To avoid aliasing of the frequency response as encountered with the impulse invariance transformation. • We need a one-to-one mapping from the s plane to the z plane. • The problem with the transformation is many-to-one. sT z e= krishnanaik.ece@gmail.com
  • 75. 75 IIR Filter Design by Bilinear transformation method • We could first use a one-to-one transformation from to , which compresses the entire s plane into the strip • Then could be transformed to z by with no effect from aliasing. s 's Im( ')s T T π π − ≤ ≤ 's 's T z e= σ jω 'σ jω /Tπ− /Tπ s domain s’ domain krishnanaik.ece@gmail.com
  • 76. 76 IIR Filter Design by Bilinear transformation method • The transformation from to is given by • The characteristic of this transformation is seen most readily from its effect on the axis. • Substituting and , we obtain s 's 12 ' tanh ( ) 2 sT s T − = jω s jω= ' 's jω= 12 ' tan ( ) 2 T T ω ω − = krishnanaik.ece@gmail.com
  • 77. 77 IIR Filter Design by Bilinear transformation method • The axis is compressed into the interval for in a one-to-one method • The relationship between and is nonlinear, but it is approximately linear at small . ( , T T π π − 'ω ω ω 'ω 'ω ω≈ - ω 'ω /Tπ /Tπ− krishnanaik.ece@gmail.com
  • 78. 78 IIR Filter Design by Bilinear transformation method • The desired transformation to is now obtained by inverting to produce • And setting , which yields 12 ' tanh ( ) 2 sT s T − = 2 ' tanh( ) 2 s T s T = s z 1 ' ( )lns z T = 2 ln tanh( ) 2 z s T = 1 1 2 1 ( ) 1 z T z − − − = + Re(Z) Im(Z) 1 S domain Z domain 1 2 1 2 T s z T s + = − jω σ krishnanaik.ece@gmail.com
  • 79. 79 IIR Filter Design by Bilinear transformation method • The discrete-time filter design is obtained from the continuous-time design by means of the bilinear transformation • Unlike the impulse invariant transformation, the bilinear transformation is one-to-one, and invertible. 1 1 (2/ )(1 )/(1 ) ( ) ( ) |c s T z z H z H s − − = − + = krishnanaik.ece@gmail.com
  • 80. 80 FIR Filter Design by Window function technique • Simplest FIR the filter design is window function technique • A supposition ideal frequency response may express where ( ) [ ]j j n d d n H e h n eω ω ∞ − =−∞ = ∑ 1 [ ] ( ) 2 j j n d dh n H e e d π ω ω π ω π − = ∫ krishnanaik.ece@gmail.com
  • 81. 81 FIR Filter Design by Window function technique • To get this kind of systematic causal FIR to be approximate, the most direct method intercepts its ideal impulse response! [ ] [ ] [ ]dh n w n h n= g ( ) ( ) ( )dH W Hω ω ω= ∗ krishnanaik.ece@gmail.com
  • 82. 82 FIR Filter Design by Window function technique • Truncation of the Fourier series produces the familiar Gibbs phenomenon • It will be manifested in , especially if is discontinuous. ( )H ω ( )dH ω krishnanaik.ece@gmail.com
  • 83. 83 FIR Filter Design by Window function technique • 1.Rectangular window • 2.Triangular window (Bartett window) 1, 0 [ ] 0, n M w n otherwise ≤ ≤ =   2 , 0 2 2[ ] 2 , 2 0, n Mn M n Mw n n M M otherwise  ≤ ≤   = − < ≤   krishnanaik.ece@gmail.com
  • 84. 84 FIR Filter Design by Window function technique • 1.Rectangular window • 2.Triangular window (Bartett window) 0 10 20 30 40 50 60 0 0.5 1 sequence (n) T(n) Rectangular window 0 10 20 30 40 50 60 0 0.5 1 sequence (n) T(n) Bartlett window 0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1 -100 -50 0 50 100 pi unitsFrequencyresponseT(jw)(dB) Rectangular window 0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1 -100 -50 0 50 100 pi units FrequencyresponseT(jw)(dB) Bartlett window krishnanaik.ece@gmail.com
  • 85. 85 FIR Filter Design by Window function technique • 3.HANN window • 4.Hamming window 1 2 1 cos , 0 [ ] 2 0, n n M w n M otherwise π   − ≤ ≤  =     2 0.54 0.46cos , 0 [ ] 0, n n M w n M otherwise π − ≤ ≤ =   krishnanaik.ece@gmail.com
  • 86. 86 0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1 -100 -50 0 50 100 pi units FrequencyresponseT(jw)(dB) Hanning window 0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1 -100 -50 0 50 100 pi units FrequencyresponseT(jw)(dB) Hamming window 0 10 20 30 40 50 60 0 0.5 1 sequence (n) T(n) Hanning window 0 10 20 30 40 50 60 0 0.5 1 sequence (n) T(n) Hamming window FIR Filter Design by Window function technique • 3.HANN window • 4.Hamming window krishnanaik.ece@gmail.com
  • 87. 87 FIR Filter Design by Window function technique • 5.Kaiser’s window • 6.Blackman window 2 0 0 2 [ 1 (1 ) ] [ ] , 0,1,..., [ ] n I Mw n n M I β β − − = = 2 4 0.42 0.5cos 0.08cos , 0 [ ] 0, n n n M w n M M otherwise π π − + ≤ ≤ =   krishnanaik.ece@gmail.com
  • 88. 88 0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1 -100 -50 0 50 100 pi units FrequencyresponseT(jw)(dB) Blackman window 0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1 -150 -100 -50 0 50 100 pi units FrequencyresponseT(jw)(dB) Kaiser window • 5.Kaiser’s window • 6.Blackman window 0 10 20 30 40 50 60 0 0.5 1 sequence (n) T(n) Blackman window 0 10 20 30 40 50 60 0 0.5 1 sequence (n) T(n) Kaiser window FIR Filter Design by Window function technique krishnanaik.ece@gmail.com
  • 89. 89 FIR Filter Design by Window function technique ( / )s Mω Window Peak sidelobe level (dB) Transition bandwidth Max. stopband ripple(dB) Rectangular -13 0.9 -21 Hann -31 3.1 -44 Hamming -41 3.3 -53 Blackman -57 5.5 -74 krishnanaik.ece@gmail.com
  • 90. 90 FIR Filter Design by Frequency sampling technique • For arbitrary, non-classical specifications of , the calculation of , n=0,1,…,M, via an appropriate approximation can be a substantial computation task. • It may be preferable to employ a design technique that utilizes specified values of directly, without the necessity of determining ' ( )dH ω ( )dh n ' ( )dH ω ( )dh n krishnanaik.ece@gmail.com
  • 91. 91 FIR Filter Design by Frequency sampling technique • We wish to derive a linear phase IIR filter with real nonzero . The impulse response must be symmetric where are real and denotes the integer part ( )h n [ /2] 0 1 2 ( 1/ 2) ( ) 2 cos( ) 1 M k k k n h n A A M π = + = + + ∑ kA [ / 2]M 0,1,...,n M= krishnanaik.ece@gmail.com
  • 92. 92 FIR Filter Design by Frequency sampling technique • It can be rewritten as where and • Therefore, it may write where 1 / 2 / 0 /2 ( ) N j k N j kn N k k k N h n A e eπ π − = ≠ = ∑ 0,1,..., 1n N= − 1N M= + k N kA A −= / 2 / ( ) j k N j kn N k kh n A e eπ π = 1 0 /2 ( ) ( ) N k k k N h n h n − = ≠ = ∑ 0,1,..., 1n N= − krishnanaik.ece@gmail.com
  • 93. 93 FIR Filter Design by Frequency sampling technique • with corresponding transform where • Hence which has a linear phase 1 0 /2 ( ) ( ) N k k k N H z H z − = ≠ = ∑ / 2 / 1 (1 ) ( ) 1 j k N N k k j k N A e z H z e z π π − − − = − ' ( 1)/2 sin / 2 ( ) sin[( / / 2)] j T N k k TN H A e k N T ω ω ω π ω − − = − krishnanaik.ece@gmail.com
  • 94. 94 FIR Filter Design by Frequency sampling technique • The magnitude response which has a maximum value at where ' sin / 2 ( ) sin[( / / 2)] k k TN H A k N T ω ω π ω = − kN A /k sk Nω ω= 2 /s Tω π= krishnanaik.ece@gmail.com
  • 95. 95 FIR Filter Design by Frequency sampling technique • The only nonzero contribution to at is from , and hence that • Therefore, by specifying the DFT samples of the desired magnitude response at the frequencies , and setting '( )H ω kω ω= ' ( )kH ω '( )k kH N Aω = ' ( )dH ω kω ' ( ) /k d kA H Nω= ± krishnanaik.ece@gmail.com
  • 96. 96 FIR Filter Design by Frequency sampling technique • We produce a filter design from equation (5.1) for which • The desired and actual magnitude responses are equal at the N frequencies ' '( ) ( )k d kH Hω ω= kω krishnanaik.ece@gmail.com
  • 97. 97 FIR Filter Design by Frequency sampling technique • In between these frequencies, is interpolated as the sum of the responses , and its magnitude does not, equal that of '( )H ω ' ( )kH ω ' ( )dH ω krishnanaik.ece@gmail.com
  • 98. 98 FIR Filter Design by Frequency sampling technique • Example: For an ideal lowpass filter from , we would choose • The frequency samples are indeed equal to the desired ' 1, 0,1,..., ( ) 0, 1,...,[ / 2] d k k P H k P M ω = =  = + ' ( ) /k d kA H Nω= ± ( 1) / ( 1), 0,1,..., 0, 1,...,[ / 2] k k M k P A k P M  − + = =  = + ' ( )kH ω ' ( )d kH ω krishnanaik.ece@gmail.com
  • 99. 99 FIR Filter Design by Frequency sampling technique • The response is very similar to the result form using the rectangular window, and the stopband is similarly disappointing. • We can try to search for the optimum value of the transition sample would quickly lead us to a value of approximately , k p≠0.38( 1) /( 1)p pA M= − + krishnanaik.ece@gmail.com
  • 100. 100 FIR Filter Design by MSE • : The spectrum of the filter we obtain • : The spectrum of the desired filter • MSE = ( )H f ( )dH f ( ) ( )∫− − − 2/ 2/ 21 s s f f ds dffHfHf 0 0.1 0.2 0.3 0.4 0.5 -0.5 0 0.5 1 1.5 krishnanaik.ece@gmail.com
  • 101. 101 FIR Filter Design by MSE • Larger MSE, but smaller maximal error • • Smaller MSE, but larger maximal error 0 0.1 0.2 0.3 0.4 -0.5 0 0.5 1 1.5 0 0.1 0.2 0.3 0.4 -0.5 0 0.5 H(F) H(F) - H (F)d 0 0.1 0.2 0.3 0.4 -0.5 0 0.5 1 1.5 0 0.1 0.2 0.3 0.4 -0.5 0 0.5 H(F) H(F) - H (F)d krishnanaik.ece@gmail.com
  • 102. 102 FIR Filter Design by MSE • 1. ( ) ( ) ( ) ( )∫∫ −− − −=−= 2/1 2/1 22/ 2/ 21 dFFHFRdffHfRfMSE d f f ds s s ( ) ( ) dFFHFnns d k n ∫ ∑− = −= 2/1 2/1 2 0 || 2cos][ π ( ) ( ) ( ) ( ) dFFHFnnsFHFnns d k n d k n ∫ ∑∑− ==       −      −= 2/1 2/1 00 2cos][2cos][ ππ ( ) ( ) 1/2 1/2 0 0 [ ]cos 2 [ ]cos 2 k k n s n n F s F dF τ π τ π τ − = = = ∑ ∑∫ ( ) ( ) ( ) 1/2 1/2 2 1/2 1/2 0 2 [ ]cos 2 k d d n s n n F H F dF H F dFπ − − = − +∑∫ ∫ krishnanaik.ece@gmail.com
  • 103. 103 FIR Filter Design by MSE • 2. when n ≠ τ, when n = τ, n ≠ 0, when n = τ, n = 0, • 3. The formula can be repressed as: ( ) ( ) 02cos2cos 2/1 2/1 =∫− dFFFn τππ ( ) ( ) 2/12cos2cos 2/1 2/1 =∫− dFFFn τππ ( ) ( ) 12cos2cos 2/1 2/1 =∫− dFFFn τππ ( ) ( ) ( )dFFHdFFHFnnsnssMSE dd k n k n ∫∫ ∑∑ −− == +−+= 2/1 2/1 22/1 2/1 01 22 2cos][22/][]0[ π krishnanaik.ece@gmail.com
  • 104. 104 FIR Filter Design by MSE • 4. Doing the partial differentiation: • 5. Minimize MSE: for all n’s ( )∫− −= ∂ ∂ 2/1 2/1 2]0[2 ]0[ dFFHs s MSE d ( ) ( )∫− −= ∂ ∂ 2/1 2/1 2cos2][ ][ dFFHFnns ns MSE dπ 0 ][ = ∂ ∂ ns MSE ( )∫− = 2/1 2/1 ]0[ dFFHs d ( ) ( )∫− = 2/1 2/1 2cos2][ dFFHFnns dπ [ ] [0] [ ] [ ]/ 2 for n=1,2,...,k [ ] [ ]/ 2 for n=1,2,...,k [ ] 0 for n<0 and n N h k s h k n s n h k n s n h n = + = − = = ≥ krishnanaik.ece@gmail.com
  • 105. 105 • IIR advantage: 1. It is easy to design 2. It is easy to implementation • IIR disadvantage: 1. Infinite impulse response 2. It is hard to optimalize than FIR 3. Non-stable krishnanaik.ece@gmail.com
  • 107. Dr. Krishnanaik VankdothDr. Krishnanaik Vankdoth B.EB.E(ECE),(ECE), M.TechM.Tech (ECE),(ECE), Ph.DPh.D (ECE)(ECE) Professor in ECE Dept Aksum University, Ethiopia– 1010 Krishnanaik.ece@gmail.com Krishnanaik_ece@yahoo.com Phone : +919441629162 krishnanaik.ece@gmail.com 107

Editor's Notes

  1. example of the time domain decomposition used in the FFT. The next step in the FFT algorithm is to find the frequency spectra of the 1 point time domain signals. Nothing could be easier; the frequency spectrum of a 1 point signal is equal to itself. This means that nothing is required to do this step. Although there is no work involved, don&amp;apos;t forget that each of the 1 point signals is now a frequency spectrum, and not a time domain signal. The last step in the FFT is to combine the N frequency spectra in the exact reverse order that the time domain decomposition took place.
  2. 问题:式中,k只有N/2个取值,只能计算X(k)的前一半的值。可利用W的周期性和对称性计算后一半的值。
  3. DIF-FFT是先做碟形运算,然后再求两个N/2点的DFT DIT-FFT是先求两个N/2点的DFT,然后再将求得的结果用碟形运算合成为一个N点的DFT。
  4. DIF-FFT是先做碟形运算,然后再求两个N/2点的DFT DIT-FFT是先求两个N/2点的DFT,然后再将求得的结果用碟形运算合成为一个N点的DFT。
  5. 有三種古典continuous
  6. because the former are monotonic in the stopband , while the latter are not
  7. The impulse invariant transformation is not usually performed directly in the form of (2.1) the parameters of H(z) may be obtained directly from H(s)
  8. 優點:To avoid aliasing of the frequency response 缺點:It is nonlinear between discrete-time frequency and continuous-time frequency.
  9. 1.main lobe越窄 resolution越高 side lobe 越低越好 2.統計上常用 resolution降一半 main lobe 變寬(trade off) Main lobe變寬(trade off) side lobe降一半
  10. 1.In fact, the length of window is M-1 2.main lobe和HANN差不多但side lobe降了10dB 3.Hamming 常用在語音處理
  11. 有參數可調,能得適當的組合