Signal Processing

Collection of Formulas
M. Dahl

S. Johansson

M. Winberg

October 26, 2005

Version 1.01

Department of Telecommunications and Signal Processing

Blekinge Institute of Technology
SE-372 33 Ronneby
Sweden

Contents
1 Basic Relationships and Concepts
1.1 Standard Signals . . . . . . . . . . . . . . . . . . . . .
1.1.1 Continuous Functions . . . . . . . . . . . . . . .
1.1.2 Discrete Time Functions . . . . . . . . . . . . .
1.2 SISO systems (Single input, single output)
Discrete Time Systems . . . . . . . . . . . . . . . . . .
1.2.1 FIR Filters . . . . . . . . . . . . . . . . . . . .
1.2.2 IIR Filters . . . . . . . . . . . . . . . . . . . . .
1.2.3 IIR Canonical form - Direct Form I . . . . . . .
1.2.4 IIR Canonical form - Direkt Form II . . . . . .
1.3 Some Methods of Calculation . . . . . . . . . . . . . .
1.3.1 Convolution . . . . . . . . . . . . . . . . . . . .
1.3.2 State-Space Model of Dierence Equation . . .
1.3.3 State-Space Equation . . . . . . . . . . . . . . .
1.3.4 System Function . . . . . . . . . . . . . . . . .
1.4 Analogue Sinusoidal Signal
through Linear, Causal Filter . . . . . . . . . . . . . .
1.4.1 Complex, Non-causal Input Signal . . . . . . . .
1.4.2 Complex, Causal Input Signal . . . . . . . . . .
1.4.3 Real, Non-causal Input Signal . . . . . . . . . .
1.4.4 Real, Causal Input Signal . . . . . . . . . . . .
1.5 Discrete Time Sinusoidal Signal
through Linear, Causal Filter . . . . . . . . . . . . . .
1.5.1 Complex, Non-Causal Input Signal . . . . . . .
1.5.2 Complex, Causal Input Signal . . . . . . . . . .
1.5.3 Real, Non-Causal Input Signal . . . . . . . . . .
1.5.4 Real, Causal Input Signal . . . . . . . . . . . .
1.6 Fourier Series Expansion . . . . . . . . . . . . . . . . .
1.6.1 Continuous Time . . . . . . . . . . . . . . . . .
1.6.2 Discrete Time . . . . . . . . . . . . . . . . . . .
1.7 Fourier Transformer . . . . . . . . . . . . . . . . . . . .
1.8 Discrete Fourier Transform (DFT) . . . . . . . . . . . .
1.8.1 Denition . . . . . . . . . . . . . . . . . . . . .
1.8.2 Circular Convolution . . . . . . . . . . . . . . .
1.8.3 Non-Circular Convolution using DFT . . . . . .
1.8.4 Relation to the Fourier Transform X(f ): . . . .
1.8.5 Relation to Fourier Series . . . . . . . . . . . .
1.8.6 Parsevals Theorem . . . . . . . . . . . . . . . .
1.9 Some Window Functions and their Fourier Transforms
2 Sampling Analogue Signals
2.1 Sampling and Reconstruction . . . . . . . . . .
2.2 Measures of Distortion . . . . . . . . . . . . . .
2.2.1 Folding distortion when sampling . . . .
2.2.2 Periodic Distortion when Reconstructing
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2.3
2.4

Quantizing Distortion . . . . . . . . . . . . . . . . . . . . . . . . . .
Decimation and Interpolation . . . . . . . . . . . . . . . . . . . . .

3 Analogue Filters
3.1 Filter Approximations if ideal LP lters . . . .
3.1.1 Butterworth Filters . . . . . . . . . . .
3.1.2 Chebyshev Filters . . . . . . . . . . . .
3.2 Frequency Transformations of Analogue Filter
4 Discrete Time Filters
4.1 FIR Filters and IIR Filters . . . . . . . . . .
4.2 Construction of FIR Filter . . . . . . . . . .
4.2.1 FIR Filter using the window method
4.2.2 Equiripple FIR Filter . . . . . . . . .
4.2.3 FIR Filters using Least-Squares . . .
4.3 Constructing IIR Filters starting from
Analogue Filters . . . . . . . . . . . . . . .
4.3.1 The Impulse-Invariant Method . . . .
4.3.2 Bilinear Transformation . . . . . . .
4.3.3 Quantizing of Coecients . . . . . .
4.4 Lattice Filters . . . . . . . . . . . . . . . . .

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5 Correlation

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6 Spectrum Estimation

39

A Basic Relationships
A.1 Trigonometrical formulas . . . . . . . . . . . . . . . . . . . . . . . .
A.2 Some Common Relationships . . . . . . . . . . . . . . . . . . . . .
A.3 Matrix Theory . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

43
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B Transforms
B.1 The Laplace Transform . . . . . . . . . . . . . . . . . . .
B.1.1 The Laplace transform of causal signals . . . . . .
B.1.2 One-sided Laplace transform of non-causal signals
B.2 The Fourier Transform of a Continuous Time Signal . . .
B.3 The Z-transform . . . . . . . . . . . . . . . . . . . . . .
B.3.1 The Z-transform of causal signals . . . . . . . . .
B.3.2 Single-Sided Z-transform of non-causal signals . .
B.4 Fourier Transform for Discrete Time Signal . . . . . . . .
B.5 Some DFT Properties . . . . . . . . . . . . . . . . . . .

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1
1.1
1.1.1

Basic Relationships and Concepts

Standard Signals
Continuous Functions

T is the period time for a perodic function f (t), where t [s] is the time variable
= 2F [rad/s], F [Hz] is the frequency.
Angular frequency = 2
T

a) Impulse Function

(t) =


+ t = 0
0 t=
 0

(t)dt = 1

x(t)(t a) = x(t a)(t)

( a1 ) = a(t)


x(t)(t a)dt = x(a)

b) Step Function

u(t) =

c) Ramp Function

r(t) =

1 t0
0 t<0
t t0
0 t<0

d) Rectangular Pulse

p(t) =

|t| <

1
2

0 |t| >

1
2

e) Sinc Function

t
sinc t = sin
t

f ) Periodic Function

f (t) = f (t + T ) where T = 0

g) Periodic Sinc Function

sin T2t
 
diric(t, T ) =
T sin 2t

h) Complex Periodic Signal

est = et ejt

i) Complex undamped per. signal

ejt = cos t + j sin t

1.1.2

Discrete Time Functions

N is the period time (integer) for a periodic function f (n), where n is a discrete
= 2f [rad], f is
time index (integer). Normalized angular frequency = 2
N
normalized frequency.
3

a) Unit Pulse

(n) =

b) Unit Step

u(n) =

c) Ramp

r(n) =

1 n=0
0 n=
 0
1 n0
0 n<0
n n0
0 n<0

1 |n| < n1

d) Rectangular Pulse

p(n) =

0 |n| > n1


e) Sinc Function

sin N2n
 
sindN (n) =
N sin n
2

f ) Periodic Signal

f (n) = f (n + N ) where N = 0

g) Complex periodic signal

esn = en ejn

h) Complex undamped per. signal

ejn = cos n + j sin n

1.2
1.2.1

SISO systems (Single input, single output)

Discrete Time Systems
FIR Filters

y(n) =

bk x(n k)

k=0

1.2.2

IIR Filters

y(n) =

ak y(n k)

k=1

1.2.3

IIR Canonical form - Direct Form I

y(n) =

ak y(n k) +

k=1

M

k=0

bk x(n k)

1.2.4

IIR Canonical form - Direkt Form II

x(n)

b0

y(n)

z -1
-a 1

b1

v N (n)
z -1
v N-1 (n)

-a 2

b2

-a N-1

bN-1

z -1
-a N

y(n) =

bN

v (n)
1

ak y(n k) +

k=1

1.3
1.3.1

Some Methods of Calculation

Convolution
y(n) = h x =

h(k)x(n k) =

k=

1.3.2

bk x(n k)

k=0

h(n k)x(k)

k=

State-Space Model of Dierence Equation

If
y(n) =

ak y(n k) +

k=1

bk x(n k)

k=0

the State-Space Model is given by

where

F=

0
..
.

v(n + 1) = Fv(n) + q x(n)

y(n) = gT v(n) + d x(n)
1
..
.

...

0
..
.

0
..
.

0
0
0
1
ak ak1 . . . a2 a1

gT = (bk , . . . , b2 , b1 ) b0 (ak , . . . , a2 , a1 ) ;

q=

0
0
..
.
1

d = b0

1.3.3

State-Space Equation

a) Direct Solution
T

y(n) = g F v(0) +
n

n1

gT Fn1k qx(k)u(n 1) + dx(n)

k=0

b) Impulse Function
h(n) = gT Fn1 qu(n 1) + d(n)
c) System Function

1.3.4

H(z) = gT [zI F]1 q + d

System Function
H(z) =

Y(z)
b0 + b1 z 1 + + bM z M
=
X (z)
1 + a1 z 1 + + aN z N

1.4

Analogue Sinusoidal Signal

through Linear, Causal Filter

1.4.1

Complex, Non-causal Input Signal

<t<

x(t) = ej0 t = (cos(0 t) + j sin(0 t))

y(t) =


=0

h( )x(t )d =


=0

h( )ej0 (t ) d = H(s)|s=j0 ej0 t





stationary
1.4.2

Complex, Causal Input Signal

x(t) = ej0 t u(t) = (cos(0 t) + j sin(0 t)) u(t);

Y (s) = H(s)X(s) =

T (s)
1
=
N (s) s j0

X(s) =

1
s j0

T1 (s)
1
+ H(s)|s=j0
N (s)
s j0


  



stationary

transient
y(t) = transient + H(s)|s=j0 ej0 t




stationary
1.4.3

Real, Non-causal Input Signal

x(t) = Re{ej0 t } = cos(0 t)
y(t) =


=0

h( )x(t )d =

<t<

1
h( ) (ej0 (t ) + ej0 (t ) )d =
2
=0

= |H(s)|s=j0 cos(0 t + arg{H(s)|s=j0 })





stationary
1.4.4

Real, Causal Input Signal

x(t) = Re{ej0 t } u(t) = cos(0 t) u(t);

Y (s) = H(s)X(s) =

T (s)
s
=
2
N (s) s + 20

X(s) =

s2

s
+ 20

T1 (s)
C1 s + C0
+ 2
N (s)
s + 20

  

transient



H(s)|s=j0 = A ej ; C1 = A cos(); C0 = A0 sin

8

stationary

C0
sin(0 t) =

 0

stationary

y(t) = transient + C1 cos(0 t) +



= transient + |H(s)|s=j0 cos(0 t + arg{H(s)|s=j0 })





stationary

1.5

Discrete Time Sinusoidal Signal

through Linear, Causal Filter

1.5.1

Complex, Non-Causal Input Signal

<n<

x(n) = ej0 n = (cos(0 n) + j sin(0 n))

y(n) =

h(k)x(n k) =

k=0

1.5.2

h(k)ej0 (nk) = H(z)|z=ej0 ej0 n



k=0



stationary

Complex, Causal Input Signal

x(n) = ej0 n u(n) = (cos(0 n) + j sin(0 n)) u(n);

Y (z) = H(z)X(z) =

T (z)
1
=
N (z) 1 ej0 z 1

X(z) =

1
1 ej0 z 1

1
T1 (z)
+ H(z)|z=ej0
N (z)
1 ej0 z 1


  
stationary
transient

y(n) = transient + H(z)|z=ej0 ej0 n





stationary
1.5.3

Real, Non-Causal Input Signal

x(n) = Re{ej0 n } = cos(0 n)

y(n) =

h(k)x(n k) =

k=0

<n<

1
h(k) (ej0 (nk) + ej0 (nk) ) =
2
k=0

= |H(z)|z=ej0 cos(0 n + arg{H(z)|z=ej0 })





stationary
1.5.4

Real, Causal Input Signal

x(n) = Re{ej0 n } u(n) = cos(0 n) u(n);

Y (z) = H(z)X(z) =

X(z) =

1 cos 0 z 1
T (z)
=
N (z) 1 2 cos 0 z 1 + z 2

1 cos 0 z 1
1 2 cos 0 z 1 + z 2

T1 (z)
C0 + C1 z 1
+
N (z)
1 2 cos 0 z 1 + z 2


  

transient
10

stationary

H(z)|z=ej0 = A ej ; C0 = A cos(); C1 = A(sin 0 sin + cos 0 cos )

y(n) = transient + C0 cos(0 n) +



C1 + C0 cos(0 )
sin(0 n) =
sin(0 )


stationary
= transient + |H(z)|z=ej0 cos(0 n + arg{H(z)|z=ej0 })




stationary

11

1.6

Fourier Series Expansion

For table see Appendix

1.6.1

Continuous Time

A periodic function with period T0 , i.e. f (t) = f (t T0 ), can be expressed in the

form of a series expansion according to
f (t) =

ck ej2kF0 t

k=

where

1 
1
ck =
f (t)ej2kF0 t dt ; F0 =
T0 T0
T0

If f (t) is real this can also be expressed as

f (t) = c0 + 2
= a0 +

|ck | cos(2kF0 t + k ) =

k=1

ak cos 2kF0 t bk sin 2kF0 t

k=1

where
1 
f (t)dt
T0 T0
2 
= 2|ck | cos k =
f (t) cos(2kF0 t)dt
T0 T0
2 
= 2|ck | sin k =
f (t) sin(2kF0 t)dt
T0 T0

a0 = c 0 =
ak
bk

The power is given by (Parsevals Relation)

P =

1 
|f (t)|2 dt =
|ck |2
T0 T0
k=

In addition, for real signals

P = c20 + 2

|ck |2 = a20 +

k=1

1.6.2

1
(a2k + b2k )
2 k=1

Discrete Time

A periodic function with the period N , i.e. f (n) = f (n N ), can be expressed as

a series expansion according to
f (n) =

N
1

ck ej2k

k=0

12

n/N

where
ck =

1
1 N
f (n) ej2k
N n=0

n/N

k = 0, . . . , N 1

The series expansion is often written using DTFS (Discrete-Time Fourier Series).
If f (n) is real this can also be expressed as


kn
f (n) = c0 + 2
|ck | cos 2
+ k =
N
k=1
= a0 +

L

k=1

where
a0 = c 0
ak = 2|ck | cos(k )
bk = 2|ck | sin(k )
N

if N even
2
L=

N 1
if N odd
2
The power is given by
P =

1
N
1

1 N
|f (n)|2 =
|ck |2
N n=0
k=0

and the energy over one period is given by

EN =

N
1

|f (n)|2 = N

n=0

1.7

N
1

|ck |2

k=0

Fourier Transformer

Continuous time signal:


j2F t

dt
Xa (F ) = xa (t)e

xa (t)

X(f )

Xa (F )ej2F t dF

Discrete time signal:

x(n)

kn
kn
ak cos 2
bk sin 2
N
N

n=

 1/2

1/2

13

x(n)ej2f n

X(f )ej2f n df



1.8

Discrete Fourier Transform (DFT)

1.8.1

Denition

Xk = DF T (xn ) =
xn = IDF T (Xk ) =
Note:

N
1

1
N

N
1

xn ej2nk/N

n=0
N
1

Xk ej2nk/N

k = 0, 1, . . . , N 1 Transform
n = 0, 1, . . . , N 1

Inversion

k=0

k0
j2 k
N n = N (k k0 , (modulo N ))
e

n=0

1.8.2

Circular Convolution
xn N y n =

N
1

DFT

x yn Xk Yk

Circular convolution

=0

where
denotes circular convolution. This means that the sequences xn and yn
should be repeated periodically before the summation. I.e. outside the interval
n = 0, 1, . . . , N 1, xnN = xn and ynN = yn ( = integer) In other words, the
index is calculated modulo N . Circular convolution is also denoted x(n) y(n).
1.8.3

Non-Circular Convolution using DFT

If x(n) = 0 for n = [0, L 1] and y(n) = 0 for n = [0, M 1] then x y = 0 for

n = [0, N 1] where N L + M 1.
The convolution can be calculated as

y = IDFT(Xk Yk ) n = 0, 1, . . . , N 1
x 

xy =

Else

where
Xk = DFT(x(n))
Yk = DFT(y(n))
1.8.4

Relation to the Fourier Transform X(f ):

X(k/N ) = Xk = DF T (x(n)) if x(n) = 0 for n = [0, N 1]

X(k/N ) = Xk = DF T (xp (n)) generally x(n) where xp (n) =

=

14

x(n N )

1.8.5

Relation to Fourier Series



k
N

= Xk = DF T (x(n)) = N ck

if
0nN 1

x(n) = xp (n),
where
xp (n) =

N
1

nk

ck ej2 N

<n<

k=0

and
ck =
1.8.6

1
nk
1 N
xp (n)ej2 N
N n=0

k = 0, 1, . . . , N 1

Parsevals Theorem
N
1

n=0

x(n)y (n) =

1
1 N
Xk Y (k)
N k=0

15

1.9

Some Window Functions and their Fourier Transforms

i) The window functions are centered around the origin (odd lter length M ),
i.e. the functions are not equal to zero only for (M 1)/2 n (M 1)/2
Rectangular window:
wrect (n) = 1
sin(f M )
M sin(f )

Wrect (f ) = M
Bartlett window (Triangular window):
w(n) = 1
M
W (f ) =
2

|n|
(M 1)/2

sin f2M
M
sin(f )
2

2

 

f
2
2

Wrect
M
2

if f is small

Hanning Window:


2n
w(n) = 0.5 + 0.5 cos
M 1

W (f ) = 0.5 Wrect (f ) +


1
+
M 1


1
+0.25 Wrect f +
M 1

+0.25 Wrect f

Hamming Window:


2n
w(n) = 0.54 + 0.46 cos
M 1

W (f ) = 0.54 Wrect (f ) +


1
+
M 1


1
+0.23 Wrect f +
M 1

+0.23 Wrect f

Blackman window:
w(n) = 0.42 + 0.5 cos

2n
4n
+ 0.08 cos
M 1
M 1

W (f ) = 0.42 Wrect (f ) +


+0.25 Wrect f
16

1
+
M 1

1
+
M 1


2
+0.04 Wrect f
+
M 1


2
+0.04 Wrect f +
M 1

+0.25 Wrect f +

Kaiser window:

 

w(n) =

I0 1 [2n/(M 1)]2
I0 ()

I0 (x) = 1 +


2
L

(x/2)k
k=1

17

k!

2
2.1

Sampling Analogue Signals

Sampling and Reconstruction

Sampling Theorem
If xa (t) is bandlimited, i.e. Xa (F ) = 0 for |F | 1/2T then

xa (t) =

x(n)

sin

n=

(t nT )
(t nT )

Sampling Frequency Fs = 1/T .

Sampling

x(n) = xa (nT ) ;


F
X(f ) = X
Fs


F
(f ) =
Fs

T =

= Fs

1
Fs

Xa (F kFs )

k=

a (F kFs )

= Fs

k=

Reconstruction (ideal)

(t nT )
(t nT )
n=
 
F
1
Fs
Xa (F ) =
X
|F |
Fs
Fs
2
 
1
Fs
F

|F |
a (F ) =
Fs
Fs
2
xa (t) =

x(n)

sin

Reconstruction using Sample-and-Hold



F
1
sin(F T ) j2F T
2 H
X

e
Xa (F ) =
LP (F )
Fs
Fs
F T
2
  
1
F  sin(F T ) 

 |HLP (F )|2

a (F ) =

Fs
Fs
F T 

18

Block Scheme describing D/A conversion

Ideal reconstruction
Lowpass
x(n)

T
-Fs/2

x(n)

S d( t-n T)
n

Fs/2

ya (t)

x(1) T

F
|X( ) T |
Fs

| X(f)|
f
-.5

y (t)
a

F
-Fs/2

.5

Fs/2

ya (t) =

|Y (F)|
a

x(n)

n=

F
1
Ya (F ) =
X
Fs
Fs

-Fs/2

sin


Fs/2

(t nT )
(t nT )
Fs
|F |
2

Reconstruction using Sample-and-Hold

Lowpass

hSH (t)

x(n)

ya (t)

1
-Fs/2

Fs/2

S d( t-n T)
n
y (t)
a

x(n)

F
|X( ) H (F)|
Fs
SH

| X(f)|
f
-.5

.5

F
-Fs/2

Ya (F ) =

|Y (F)|
a

Fs/2

F
1
sin(F T ) j2F
e
X

Fs
Fs
F T
19

-Fs/2

T
2

HLP (F )

Fs/2

2.2
2.2.1

Measures of Distortion
Folding distortion when sampling

Spectrum after anti-aliasing lter:

in (F )
Folding distortion:
DA = 2
Eective signal power:
Ds = 2


Fs Fp

 Fp
0

in (F )dF

in (F )dF

where 0 Fp Fs /2
Signal distortion relationship:
A: SDRA =

 Fp
in (F )dF

= 0

DS
DA

Fs Fp

0
B: SDRA
= min|F |Fp

in (F )dF

in (F )
in (Fs F )

If the spectrum is monotonously decreasing

0
=
SDRA

2.2.2

in (Fp )
in (Fs Fp )

Periodic Distortion when Reconstructing

Periodic Distortion:
DP = 2
Eective signal power:
DS = 2


Fs /2

 Fs /2
0

ut (F )dF

ut (F )dF

Signal Distortion Relationship:

A:

SDRP =

DS
DP

 Fs /2
ut (F )dF
= 0
Fs /2

B: SDRP0 = min|F |<Fs /2

ut (F )dF

ut (F )
ut (Fs F )

A good estimate is often given by

SDRP0 =

ut (Fp )
ut (Fs Fp )

where Fp is the highest frequency component in the sampled signal.

20

2.3

Quantizing Distortion
2
DQ

linear quantizing, small

12
SQNR =

Signal Power
Dq

Quantizing distortion for sinusoidal signal, maximum dynamical range used, r bits
SQNR = 1.76 + 6 r[dB]
Quantizing distortion, dynamical range usage expressed in top- and RMS value, r
bits


Apeak

SQNR = 6 r + 1.76 1010 log

ARM S 2

2

V
1010 log
Apeak

where [V, V ] is the dynamical range of the quantizer.

2.4

Decimation and Interpolation

Downsampling a factor M
M

y(n) = {. . . u(0), u(M ), u(2M ) . . .}

Y (f ) =

1
M

M 1
i=0

f i
M

Upsampling a factor L
L

w(n) = {. . . x(0), 0, 0, . . ., x(1), 0, 0, . . ., x(2) . . .}

  

  

L-1 st

L-1 st

W (f ) = X(f L)

21

2

Analogue Filters

3.1

Filter Approximations if ideal LP lters

General form of the amplitude function of the approximation

K

|H()| = 

1 + gN
where



= 2F

2 

 

  < 1

2
1

gN

1


 

  > 1
p

and p is the cut-o frequency of the lter.

Sometimes it is suitable to normalize the angular frequency to p .
In this section this corresponds to setting p = 1.
3.1.1

Butterworth Filters
K

|H()| = 
1+

2N

The amplitude function

always passes these points.

K = The maximum value of the amplitude function.

K = The value of the amplitude functionens for = 0.

Arbitrary attenuation in the pass band (X dB)

K

|H()| = 

1+


When p = p (3dB) then 2 1.

22


p

2N

The amplitude function

always passes these points.
Any arbitrary
attenuation

Filter order:
N=

log10

 

1
22

log10

1 /
s
p

2 is the maximum allowed value for the amplitude function at the stop band edge:
2 = |H(s )|max
The denominator of the system function is a Butterworth polynomial if p = 1.
These polynomials can be found in Table 2.1. For a general p
H(s) = 

s
p

N

+ aN 1

s
p

N 1

where a1 , . . . , aN 1 can be found in Table 4.1.

23

+ + a1

s
p

+1

Table 4.1
Coecients a in Butterworth polynomials sN + aN 1 sN 1 + . . . + a1 s + 1
N
1
2
3
4
5
6
7
8

a1

a2

a3

a4

a5

a6

a7

2
3.414
5.236
7.464
10.103
13.138

2.613
5.236
9.141
14.606
21.848

3.236
7.464
14.606
25.691

3.864
10.103
21.848

4.494
13.138

5.126

2
2
2.613
3.236
3.864
4.494
5.126

Table 4.2
Factorized Butterworth polynomials for p = 1. For p = 1 let s s/p .
N
1
2
3
4
5
6
7
8
3.1.2

(s + 1)
(s2 + 2s + 1)
(s2 + s + 1)(s + 1)
(s2 + 0.76536s + 1)(s2 + 1.84776s + 1)
2
(s + 1)(s2 + 0.6180s + 1)(s
+ 1.6180s + 1)

(s2 + 0.5176s + 1)(s2 + 2s + 1)(s2 + 1.9318s + 1)

(s + 1)(s2 + 0.4450s + 1)(s2 + 1.2465s + 1)(s2 + 1.8022s + 1)
(s2 + 0.3896s + 1)(s2 + 1.1110s + 1)(s2 + 1.6630s + 1)(s2 + 1.9622s + 1)
Chebyshev Filters
|H()| = 

|H( )|

Ripple

K
K
1+ 2

K
1 + 2 TN2

N odd

The amplitude function

always passes this point.

N even
p
Not necessary the 3 dB frequency

Ripple = 10 log(1 + 2 ) dB.

K = The maximum value of the amplitude function.
24

K = the value of the amplitude function for = 0 when

 N is even.

TN ( p ) is a Chebyshev polynomial. (Also denoted CN p ). These can be found

in Table 4.3 for p = 1. For p = 1 let p in Table 4.3.
System function

K a0
H(s) = 

s
p

N

N odd
N even

1
1+2

+ aN 1

s
p

N 1

+ + a0

where , a0 , . . . , aN 1 can be found in Table 4.4.

The locations of the poles for H(s) can be found in Table 4.5 for p = 1. For
p = 1 the pole locations are multiplied with p .
Table 4.3
Chebyshev polynomials.

cos(N arccos )

TN () =

or

TN () =

|| 1
= 2F

cosh(N arccosh) || 1

N

2 1

+ +

N

2 1

Recursive calculation
TN +1 () = 2TN () TN 1 ()

Filter order:
arccosh
N=

 

1
22

arccosh

1 /
s
p

2 is the maximum allowed stop band ripple.

Alternative for arccosh:



arccosh(x) = ln x +

25

x2 1

|| 1

N
0
1
2
3
4
5
6
7
8
9
10

TN ()
1

22 1
43 3
84 82 + 1
165 203 + 5
326 484 + 182 1
647 1125 + 563 7
1288 2566 + 1604 322 + 1
2569 5767 + 4325 1203 + 9
51210 12808 + 11206 4004 + 502 1

26

Table 4.4. Chebyshev lter coecients a .

0.5dB ripple ( = 0.349, 2 = 0.122).
N
a7
a6
a5
a4
a3
a2
1
2
3
1.253
4
1.197 1.717
5
1.172 1.937 1.309
6
1.159 2.172 1.589 1.172
7
1.151 2.413 1.869 1.648 0.756
8 1.146 2.657 2.149 2.184 1.148 0.573
1-dB ripple ( = 0.509, 2 = 0.259).
N
a7
a6
a5
a4
a3
a2
1
2
3
0.989
4
0.953 1.454
5
0.937 1.689 0.974
6
0.928 1.931 1.202 0.939
7
0.923 2.176 1.429 1.357 0.549
8 0.920 2.423 1.655 1.837 0.447 0.448
2-dB ripple ( = 0.765, 2 = 0.585).
N
a7
a6
a5
a4
a3
a2
1
2
3
0.738
4
0.716 1.256
5
0.705 1.499 0.693
6
0.701 1.745 0.867 0.771
7
0.698 1.994 1.039 1.144 0.383
8 0.696 2.242 1.212 1.579 0.598 0.359
3-dB) ripple ( = 0.998, 2 = 0.995).
N
a7
a6
a5
a4
a3
a2
1
2
3
0.597
4
0.581 1.169
5
0.575 1.415 0.549
6
0.571 1.663 0.691 0.699
7
0.568 1.911 0.831 1.052 0.300
8 0.567 2.161 0.972 1.467 0.472 0.321
*) The table was calculated with exactly 3dB,
Hence = 1 and a0 = 1 for N = 1.

27

a1
1.426
1.535
1.025
0.752
0.432
0.282
0.152
a1
1.098
1.238
0.743
0.580
0.307
0.214
0.107
a1
0.804
1.022
0.517
0.459
0.210
0.166
0.073
a1

a0
2.863
1.516
0.716
0.379
0.179
0.095
0.045
0.024
a0
1.965
1.102
0.491
0.276
0.123
0.069
0.031
0.017
a0
1.307
0.823
0.327
0.206
0.082
0.051
0.020
0.013

a0
1.002
0.645 0.708
0.928 0.251
0.405 0.177
0.408 0.063
0.163 0.044
0.146 0.016
0.056 0.011

not with 20 log 2 3.01dB.

Table 4.5. Pole locations for Chebyshev lters.

0.5dB ripple ( = 0.349, 2 = 0.122).
N =1
2
3
4
5
6
7
-2.863
-0.713
-0.626
-0.175
-0.362
-0.078
-0.256
j1.004
j1.016
j1.008
-0.313
-0.423
-0.112
-0.212
-0.057
j1.022 j0.421 j1.011 j0.738 j1.006
-0.293
-0.290 0.160
j0.625 j0.270 j0.807
-0.231
j0.448
1-dB ripple ( = 0.509, 2 = 0.259).
N =1
2
3
4
5
6
7
-1.965
-0.549
-0.494
-0.139
-0.289
-0.062
-0.205
j0.895
j0.983
j0.993
-0.247
-0.337
-0.089
-0.170
-0.046
j0.966 j0.407 j0.990 j0.727 j0.995
-0.234
-0.232
-0.128
j0.612 j0.266 j0.798
-0.185
j0.443
2
2-dB ripple ( = 0.765, = 0.585).
N =1
2
3
4
5
6
7
-1.307
-0.402
-0.369
-0.105
-0.218
-0.047
-0.155
j0.813
j0.958
j0.982
-0.184
-0.253
-0.067
-0.128
-0.034
j0.923 0.397 j0.973 0.719 j0.987
-0.177
-0.175
-0.097
j0.602 j0.263 j0.791
-0.140
j0.439
3-dB) ripple ( = 0.998, 2 = 0.995).
N =1
2
3
4
5
6
7
-1.002
-0.322
-0.299
-0.085
-0.177
-0.038
-0.126
j0.777
j0.946
j0.976
-0.1493
-0.206
-0.055
-0.104
-0.028
j0.904 j0.392 j0.966 0.715 j0.983
-0.144
-0.143
-0.079
j0.597 j0.262 j0.789
-0.114
j0.437
*) See note in Table 4.4.

28

8
-0.044
j1.005
-0.124
j0.852
-0.186
j0.570
-0.220
j0.200
8
-0.035
j0.996
-0.100
j0.845
-0.149
j0.564
-0.176
j0.198
8
-0.026
j0.990
-0.075
j0.839
-0.113
j0.561
-0.133
j0.197
8
-0.021
0.987
-0.061
j0.836
-0.092
j0.559
-0.108
j0.196

3.2

Frequency Transformations of Analogue Filter

1. Start out from the frequencies in the lter specication in the analogue high
pass-, band pass- or band stop lter.


2. Transform to the frequencies of the LP lter 2 and 3 .

3. Find the coecients of the LP lter coecients.
4. Transform back to the original lter (HP, BP, BS) by replacing s i the low
pass lter H(s); see below.

LP
()

HP

BP

()

2 3

()

3 2

HP LP 2 =

BS LP

2 =

LP

3 =

2 = 2 1

()

3 l 2 4

1
2

BP LP

BS

1
3

1 2 = 3 4 = 2I

3 = 4 3

2I
4 3

3 =

2I
2 1

1 2 = 3 4 = 2I

HP

BP

s + sI

29

3 l 2 4

BS
2I

s + sI

4
4.1

Discrete Time Filters

FIR Filters and IIR Filters
H(z) = b0 + b1 z 1 + . . . + bM 1 z M +1


bn 0 n M 1
h(n) =
0 Else

Linear phase FIR Filter :

Symmetrical impulse response h(n) = h(M 1 n)
Anti-symmetrical impulse response h(n) = h(M 1 n)

IIR Filter
b0 + b1 z 1 + . . . + bM z M
1 + a1 z 1 + . . . + aN z N
h(n) = Z 1 {H(z)}

H(z) =

4.2
4.2.1

Construction of FIR Filter

FIR Filter using the window method

Impulse response
h(n) = hd (n) w(n)
with desired impulse response hd (n) and time window w(n).
Filters designed using the window method have linear phase.
Desired impulse response hd (n) dened for 0 n M 1
Odd lter length
Low pass:


M 1
c sin c n 2


hd (n) =
c n M 1

hd (

c
M 1
) =
2


ej
Hd () =
0

(M 1)/2

0 || < c
Else

High pass:


M 1

M 1
c sin c n 2


hd (n) = n

2
c n M 1
2

Hd () =

ej
0

(M 1)/2

30

c < || <
Else

Band pass:


hd (n) = 2 cos 0

Hd () =

ej
0

M 1
n
2

(M 1)/2



M 1
c sin c n 2



c n M 1
2

0 c < || < 0 + c
Else

Band stop:


M 1
M 1
hd (n) = n
2 cos 0 n
2
2

Hd () =

ej
0

(M 1)/2



M 1
c sin c n 2



c n M 1
2

0 < || < 0 c and 0 + c < || <

Else

The spectrum of the lter H() = Hd () W ().

At the cut-o frequency c the attenuation is 6dB.
When dimensioning lters, the tables below oers a rough estimation of the required
lter length M .
Table 5.1
The main and sidelobe size for some common window functions.
Approximate
Window
main lobe width
(rad)
Rektangular
4/M
Bartlett
8/M
Hanning
8/M
Hamming
8/M
Blackman
12/M

Largest sidelobe
relative main lobe
(dB)
-13
-27
-32
-43
-58

Table 5.2
Sidelobe size for some lters designed using the window method.
Approximate
Window
largest sidelobe
(dB)
Rektangulr
-20
Bartlett
-27
Hanning
-40
Hamming
-50
Blackman
-70
Kaiser ( = 4.54)
-50
Kaiser ( = 6.76)
-70
Kaiser ( = 8.96)
-90
31

Window functions dened for 0 n M 1 (Odd lter length M )

Rectangular window
w(n) = 1
Bartlett (Triangular window)
w(n) = 1





n M21 
M 1
2

Hanning window


2 n

w(n) = 0.5 1 + cos



= 0.5

M 1
2

M 1
2n
M 1

1 cos

Hamming window


w(n) = 0.54 + 0.46 cos

2 n

M 1
2

M 1
2n
= 0.54 0.46 cos
M 1

Blackman window


w(n) = 0.42 + 0.5 cos

2 n

4 n

= 0.42 0.5 cos

M 1

+0.08 cos

M 1
2

M 1
2

M 1

2n
4n
+ 0.08 cos
M 1
M 1

Kaiser window
 

w(n) =

I0 1 [2(n

M 1
)/(M
2

1)]2

I0 ()

I0 (x) is a Bessel function, whose value can be calculated according to

I0 (x) = 1 +


2
L

(x/2)k
k=1

k!

Usually L < 25.

The value of is decided from the stop band attenuation D.

32

Stop band attenuation D

D 21dB
0
21dB < D < 50dB
0, 5842(D 21)0,4 + 0, 07886(D 21)
D 50dB
0, 1102(D 8, 7)
Filter length M :
M

D 7, 95
14, 36f

f = fs fp
fp Pass band frequency (normalized frequency)
fs Stop band frequency (normalized frequency)

33

4.2.2

Equiripple FIR Filter

Dimensioning of equiripple lters when using the Remez algorithm. Approximately

according to Kaiser.

|H(f)|
1+ d p
1
1- d p
ds

f
fp

Filter length M =

fs

20 log10



p s 13

14.6f

+1

f = fs fp

4.2.3

FIR Filters using Least-Squares

Minimizing
E=

[x(n) h(n) d(n)]2

yields

M
1

h(n)rxx (n ) = rdx ()  = 0, . . . , M 1

n=0

and
Emin = rdd (0)

M
1

h(k)rdx (k)

k=0

where rxx () is the correlation function for x(n) and rdx () is the cross correlation
between d(n) and x(n).
This can be written on matrix form as
Rxx h = rdx
h = R1
xx rdx
Emin = rdd (0) hT rdx

34

4.3
4.3.1

Constructing IIR Filters starting from

Analogue Filters
The Impulse-Invariant Method
h(n) = ha (t)|t=nT = ha (nT )

H(z) =

ha (nT )z n

n=0

1.
ha (t) = e0 t Ha (s) =

1
s + 0

H(z) =

e0 T z 1

2.
ha (t) = e0 t cos 0 t Ha (s) =

s + 0
(s + 0 )2 + 20

1 z 1 e0 T cos 0 T
H(z) =
1 2z 1 e0 T cos 0 T + z 2 e20 T
3.
ha (t) = e0 t sin 0 t Ha (s) =
H(z) =
4.3.2

0
(s + 0 )2 + 20

z 1 e0 T sin 0 T
1 2z 1 e0 T cos 0 T + z 2 e20 T

Bilinear Transformation

Frequency transformation (prewarp)

prewarp =

tan
T
2

Analogue construction of lters in the variable prewarp = 2Fprewarp

.
2 1 z 1
H(z) = Ha (s) where s =
T 1 + z 1
T is a normalization factor (can often be set to 1).

35

( = 2f )

4.3.3

Quantizing of Coecients

Change in pole locations when the coecients a1 , . . . , ak is changed a1 , . . . , ak

pi

pi
pi
a1 + +
ak
a1
ak

If using normal form (Direct form II) then

pi
pkj
i
=
aj
(pi p1 )(pi p2 ) . . . (pi pk )




k1

factors

(pi pi ) should not be included

4.4

Lattice Filters
f0 (n)

f1 (n)

K1

x(n)

g0(n)

...

K2
K1

z -1

f2(n)

g1(n)

K
K2

z -1

fM-1(n) = y(n)

...

z -1

g2(n)

+
M

(n)
-1

Lattice FIR

x(n)
f N(n)

+
-

f N-1(n)

f (n)
... 2

f 1(n)

+
-

KN

z -1

...

f N(n)

gN (n)

+
-

f N-1(n)
KN
KN

z -1

K1

+
z -1
g2 (n)
Lattice all-pole IIR

f (n)
... 2
-

g0(n)

f0(n)

+
K1

K2
z -1

g1 (n)

K1
+

g0(n)
v0

z -1

v1

v2
...

z -1

K2

g (n)
... 2
+

vN

g1 (n)

f 1(n)

+
-

f0(n) = y(n)
K1

K2

gN (n)

x(n)

K2

KN
+

y(n)

+
Lattice-ladder

36

Conversion from Lattice to Direct form:

A0 (z) = B0 (z) = 1

Am (z) = Am1 (z) + Km z 1 Bm1 (z)

Bm (z) = Km Am1 (z) + z 1 Bm1 (z)
m = 1, 2, . . . , M 1

Relationship between Am (z) and Bm (z)

Am (z) = m (0) + m (1)z 1 + ... + m (m 1)z m+1 + m (m)z m
Bm (z) = m (0) + m (1)z 1 + ... + m (m 1)z m+1 + m (m)z m
Bm (z) = z m Am (z 1 )
m (k) = m (m k)
k = 0, 1, . . . , m
Conversion from Direct form to Lattice:
1
(Am (z) Km Bm (z))
Am1 (z) =
2
1 Km
m = M 1, M 2, . . . , 1
Reection coecient:
Km = m (m)
FIR Filter
H(z) = AN (z)
AM 1 (z) = AN (z)
IIR Filter (all-pole)
H(z) =

1
AN (z)

AM 1 (z) = AN (z)
Lattice-Ladder
Conversion from Direct form to Lattice-Ladder:
H(z) =

CM (z)
c0 + c1 z 1 . . . cM z M
=
AN (z)
AN (z)

Cm1 (z) = Cm (z) vm Bm (z)

vm = cm (m) m = 0, 1, . . . , M
37

Correlation

Correlation, Cross Correlation, spectrum, cross spectrum and coherence between

input and output signal.
Continuous time process:
y(t)
Y (F )
ryy ( )
Ryy (F )
ryx ( )
Ryx (F )
rxx ( )
ryx ( )
xx ( )
yx ( )

=
=
=
=
=
=
=
=
=
=

h(t) x(t)
H(F ) X(F )
rhh ( ) rxx ( )
|H(F )|2 Rxx (F )
h( ) rxx ( )
H(F
) Rxx (F )

x(t)x(t )dt
t
t y(t)x(t )dt
E{x(t)x(t )}
E{y(t)x(t )}

y(n)
Y (f )
ryy (n)
Ryy (f )
ryx (n)
Ryx (f )
rxx (n)
ryx (n)
xx (n)
yx (n)

=
=
=
=
=
=
=
=
=
=

h(n) x(n)
H(f ) X(f )
rhh (n) rxx (n)
|H(f )|2 Rxx (f )
h(n) rxx (n)
H(f ) Rxx (f )

x()x( n)

 y()x( n)
E{x()x( n)}
E{y()x( n)}

Discrete time process:

Normally distributed stochastic variables. Xi N (mi , i )

E{X1 X2 X3 X4 } = E{X1 X2 } E{X3 X4 } + E{X1 X3 } E{X2 X4 } +
+ E{X1 X4 } E{X2 X3 } 2m1 m2 m3 m4

38

Spectrum Estimation

Spectrum Estimation
xx (m) = E{x(n)x(n + m)} autocorrelation

xx (f ) =

xx ej2f m power spectrum

m=

1
rxx (m) =
N
Pxx (f ) =

N m1

x(n)x(n + m) 0 m N 1 auto correlation (estimate)

n=0

N
1

m=N +1

rxx (m)e

j2f m

1
=
N


2
1
N


j2f m 


x(m)e



power spectrum (estimate)

m=0

Periodogram

1
Pxx (f ) =
N

E{rxx (m)} =


2
1
N


j2f m 


x(m)e



power spectrum (estimate)

m=0

|m|
1
xx (m) xx (m) when N
N

1
2
[xx
(n) + xx (n m)xx (n + m)] 0 when N
var(rxx (m))
N n=

E{Pxx (f )} =

 1/2
1/2

xx ()WB (f )d


where WB (f ) is the Fourier transform of the Bartlett window 1

var(Pxx (f )) = 2xx (f ) 1 +

sin2f N
N sin2f

2

Periodogram using DFT:

Pxx

k
N

1
=
N

N 1



x(n)ej2

n=0

39

2 (f ) when N
xx

if x(n) Gaussian.

|m|
N

nk
N

2





k = 0, . . . , N 1

Bartletts method
Averaging periodograms.
Divide the N sample data sequence x(n) into K blocks with M samples in each
block. The blocks are not overlapping (N = K M ).
xi (n) = x(n + iM ) i = 0, 1 . . . , K 1 and n = 0, 1 . . . , M 1

1
i
Pxx
(f ) =
M
B
(f ) =
Pxx

N 1
2



j2f m 


x
i (m)e



i = 0, 1 . . . , K 1

m=0

1 K1
P i (f )
K i=0 xx

1 K1
i
i
E{Pxx
(f )} = E{Pxx
(f )}
K i=0
1  1/2
i
E{Pxx (f )} =
xx ()WB (f )d
M 1/2

B
E{Pxx
(f )} =

where WB (f ) is the Fourier Transform of the Bartlett window 1

1
WB (f ) =
M

B
var(Pxx
(f )) =

sin f M
sin f

2

1 K1
i
var(Pxx
(f ))}
2
K i=0

B
(f ))
var(Pxx

|m|
N

1 2
sin2f N
=
xx (f ) 1 +
K
N sin2f

2

Welchs method
Averaging modied periodograms.
Divide the N sample data sequence x(n) into L block with M samples in each
block. The blocks may be overlapping. The starting point for block i is given by
iD. D = M means no block overlap and D = M/2 yields 50% overlap.
xi (n) = x(n + iD) i = 0, 1 . . . , L 1 and n = 0, 1 . . . , M 1

40

Modied periodogram.
i
Pxx
(f )

1
=
MU

w(n)


2
1
N


j2f m 


x
(m)w(n)e
i



i = 0, 1 . . . , L 1

m=0

window function

U =

1
1 N
w2 (n)
M n=0

W
(f ) =
Pxx

1 L1
P i (f )
L i=0 xx

W
(f )} =
E{Pxx
i
(f )} =
E{Pxx

1 L1
i
i
E{Pxx
(f )} = E{Pxx
(f )}
L i=0

 1/2

1/2

1
W (f ) =
MU

xx ()W (f )d

M 1
2




w(n)ej2f n 


n=0

No block overlap (L = K)
W
var(Pxx
(f )) =

1
i
(f ))}
var(Pxx
L
1 2
(f )
L xx

50% block overlap (L = 2K) and Bartlett window (triangular window).

W
(f ))
var(Pxx

41

9 2
(f )
8L xx

Averaging Periodograms
Quality factor
Q=

[E{Pxx (f )}]2
var(Pxx (f ))

Relative variance Q1
Q time-bandwidth product.
f =

0.9
M

Q=1

Bartlett (N = K M ) f =

0.9
M

Q=

N
M

Welch (N = L M )

f =

1.28
M

Q=

16
9

Welch (N = L M )

f =

1.50
M

Q=

3
1.082

Welch

f =

1.50
M

Q=

3
2

N
M

Hanning window
62,5% overlap

Blackman/Tukey

f =

0.6
M

Q=

1
2

N
M

Rectangular window

f =

0.9
M

Q=

3
2

N
M

Triangular window

Periodogram

Rectangular window
No overlap

N
M

Triangular window
50% overlap
N
M

Hanning window
50% overlap

The resolution f is calculated at the -3dB points from the main lobe of the
window.

42

A
A.1

Basic Relationships
Trigonometrical formulas
sin = cos( /2)
cos = sin( + /2)
cos2 + sin2 = 1
cos2 sin2 = cos 2
2 sin cos = sin 2
sin() = sin
cos() = cos
cos2 = 12 (1 + cos 2)
sin( + ) = sin cos + cos sin
cos( + ) = cos cos sin sin
2 sin sin = cos( ) cos( + )
2 sin cos = sin( + ) + sin( )
2 cos cos = cos( + ) + cos( )
cos
sin + sin = 2 sin +
2
2

cos + cos = 2 cos +

cos
2
2
cos =

1
2

(ej + ej ), sin =

1
2j

A cos + B sin =
where cos =

(ej ej ), ej = cos + j sin

A2 + B 2 cos( )

A
,
A2 +B 2

sin =

arctan A

and =

if A 0

arctan B
+
A

A cos + B sin =
where cos =

B
,
A2 +B 2

sin =

A
arctan B
+

43

if A < 0

A2 + B 2 sin( + )

arctan B

and =

B
A2 +B 2

A
A2 +B 2

if B 0
if B < 0

Degrees Rad
0

30
45
60
90

A.2

sin

cos

tan

cot

0
1
2
1

2
3
2
1

0
1

3 1
3
2
3
1
1
1
2

1
3 1
2
3
0 0

Some Common Relationships

Sum of geometrical series

N
1

an =

n=0

1aN
1a

if a = 1
if a = 1

Summation of sinusoid over an even number of periods

N
1

j2 kn/N

n=0

A.3

N if k = 0, N, . . .
0 Else

Matrix Theory

Matrix notation A and vector x

A matrix A with dimension m n and a vector x with dimension n is dened by

A=

a11
a21
..
.

a12
a22
..
.

. . . a1n
. . . a2n
..
.

x=

am1 am2 . . . amn

The matrix A is symmetrical if aij = aji ij.
I denotes the unity matrix.
Transposing matrix A
B = AT where bij = aji
(AB)T = BT AT

44

x1
x2
..
.
xn

Matrix A determinant

 a11


 a21
det A = |A| =  ..
 .

 am1

a12
a22
..
.





n


=
aij (1)i+j det Mij


i=1

amn 

. . . a1n
. . . a2n
..
.

am2 . . .

where Mij is the resulting matrix if row i and column j in matrix A is deleted.
det AB = det A det B
Especially, for a 22 matrix:
det A = a11 a22 a12 a21
Inverse of matrix A
A1 A = AA1 = I (if det A#0)
A1 =

1
CT
det A

where C is dened by
cij = (1)i+j det Mij
(AB)1 = B1 A1
Especially for a 22 matrix:
A1

1
=
det A

a22 a12
a21
a11

Eigenvalues and Eigenvectors

The eigenvalues (i , i = 1, 2, . . . , n) and the eigenvectors (qi , i = 1, 2, . . . , n) are
solutions for the equation system
Aq = q or (A I)q = 0
The eigenvalues can be calculated as solutions to the characteristic equation (secular equation) for A
det(I A) = n + n1 n1 + + 0 = 0
det(I A) is called the characteristic polynomial (the secular polynomial) for A.

45

Transforms

B.1
B.1.1

The Laplace Transform

The Laplace transform of causal signals

In the table below f (t) = 0 for t < 0 (i.e. f (t) u(t) = f (t)).
1. f (t) =
1
2j

2.

 +j
j

F(s)e ds
st

a f (t)

3. f (at)
4.

1
a

F(s) =

st
dt
0 f (t)e


1
a

a F (s) Linearity

F( as ) Scaling

F(as) a > 0 Scaling

f ( at )

5. f (t t0 ); t t0

F(s) est0 Time shift

6. f (t) eat

F(s + a) Frequency shift

7.

dn f
dtn

t

sn F(s) Derivation
f ( )d

1
s

9. (t)n f (t)

dn F (s)
dsn

8.

10.

f (t)
t

F(s) Integration

Derivation in
frequency domain


F(z)dz Integration in
frequency domain
s

11. limt0 f (t) = lims s F(s)

Initial value
theorem

12. limt f (t) = lims0 s F(s)

End value theorem

13. f1 (t) f2 (t) =

t
f ( )f2 (t )d =
0t 1
0 f1 (t )f2 ( )d

F1 (s) F2 (s)
Convolution in time domain

14. f1 (t) f2 (t)

15.

f1 (t) f2 (t)dt
0 
+j
1
F1 (s)
2j j

=
F2 (s)ds

1
2j
1
2j

F1 (s) F2 (s) =
j F1 (z) F2 (s z) dz
Convolution in frequency domain
 +j

Parsevals relation

46

16. (t)

17. n (t)

sn

18. 1

1s

1 tn
19. n!

20. e0 t

s +1
0

21.

B.1.2

1
sn+1

1
1
tn1 e0 t
(n 1)!
(s + 0 )n

22. sin 0 t

0
s2 + 20

23. cos 0 t

s
s2 + 20

24. t sin 0 t

20 s
(s2 + 20 )2

25. t cos 0 t

s2 20
(s2 + 20 )2

26. e0 t sin 0 t

0
(s + 0 )2 + 20

27. e0 t cos 0 t

s + 0
(s + 0 )2 + 20

28. e0 t sin(0 t + )

(s + 0 ) sin + 0 cos
2
2
(s + 0 ) + 0

One-sided Laplace transform of non-causal signals

Notation

F + (s) = 0
f (t)est dt

Single sided Laplace transform,

f (t) not necessarily causal.
For causal signals

F(s) = F + (s)
Taking the derivative of f (t) yields
d
f (t)
dt

s F + (s) f (0)

dn
f (t)
dt

sn F + (s) sn1 f (0)

Single derivation

sn2 f (1) (0) . . . f (n1) (0) n derivations

47

B.2

The Fourier Transform of a Continuous Time Signal

= 2F

1. w(t) = F 1 {W (F )} =

j2F t
dF
W (F )e
2.

W (F ) = F{w(t)} =

j2F t
dt
w(t)e

a w (t)

a W (F )

3. w (t)

W (F )

4. W (t)

w(F )

5. w(at)

6. w(t t0 )

W (F ) ej2F t0

7. w(t) ej2F0 t

W (F F0 )

8. w (t)

W (F )

9.

dn w(t)
dtn

t

1
|a|

 
F
a

(j2F )n W (F )

w( )d

1
j2F

11. j2t w(t)

dw
dF

12. w1 (t) w2 (t)

W1 (F ) W2 (F )

13. w1 (t) w2 (t)

W1 (F ) W2 (F )

10.

14.

|w(t)|2 dt =

|W (F )|2 dF

W (F ) if W (F ) = 0 for F = 0

Parsevals relation

15. 
w1 (t) w2 (t)dt =

W1 (F ) W2 (F )dF

w1 (t), w2 (t) real

16. (t)

17. 1

(F )

18. u(t)

1
j2F

19. eat u(t)

1
a+j

48

1
2

(F )

2a
a2 +2

20. ea|t|

21. ej2F0 t

(F F0 )

22. sin 2F0 t

23. sin 2F0 t u(t)

0
20 2

24. cos 2F0 t

1
2

25. cos 2F0 t u(t)

j
20 2

26.

1
2 2

et

2 /2 2

1
2

{(F + F0 ) (F F0 )}
+j

1
4

{(F + F0 ) (F F0 )}

{(F + F0 ) + (F F0 )}

e()

+ 14 {(F + F0 ) + (F F0 )}
2 /2

27. eat sin 2F0 t u(t)

0
(j + a)2 + (0 )2

28. ea|t| sin 2F0 |t|

20 (20 + a2 2 )
(2 + a2 20 )2 + 4a2 20

29. eat cos 2F0 t u(t)

j + a
(j + a)2 + (0 )2

30. ea|t| cos 2F0 t

2a(20 + a2 + 2 )
( + a2 20 )2 + 4a2 20

1
a

sinc( Fa ) a > 0

1
a

rect( Fa ) a > 0

31. rect(at) =
32. sinc(at) =

1 for |t| <

0 Else

1
2a

sin(at)
at

33. repT (w(t)) =

m=

w(t mT )

34. |T |combT (w(t)) =


|T |
m= w(mT )(t mT )
35.

n= cn (t

nT )

1
|T |

comb1/T (W (F ))

rep1/T (W (F ))

49

1
n= T

cn (F Tn ) =

cn ej2nT F

B.3

The Z-transform

B.3.1

The Z-transform of causal signals



1. X (z) = Z[x(n)] =

n=
1
2j

2. x(n) = Z 1 [X (z)] =
3.

a x (n)

x(n)z n

Transform

X (z)z n1 dz

Invers transform

a X (z)

Linearity

4. x(n n0 ) z n0 X (z)

Shift (n0 positive or

negative integer)

5. nx(n) z

Multiplication with n

6. an x(n) X
7. x(n) X
8.

(

n
=

d
dz

X (z)

 
z
a

Scaling

 
1
z

Reection of the time sequence

x()

z
z1

X (z)

Summation

9. x y X (z) Y(z)
10. x(n) y(n)

1
2j

Convolution

Y()X

 
z

1 d

Product

11. x(0) = limz X (z) (if the limit value exist) Initial value theorem
12. limn x(n) = limz1 (z 1)X (z)
(if the unit cirle is included in the ROC)
13.

14.

= x()y() =

=

x2 () =

1
2j

1
2j

x(z)y

 
1
z

z 1 dz

X (z)X (z 1 )z 1 dz

50

End value theorem

Parsevals teorem for

real time sequences

Sequence

Transform

x(n)

X (z)

15. (n)

16. u(n)

1
1 z 1

17. nu(n)

z 1
(1 z 1 )2

18. n u(n)

1
1 z 1

19. (n + 1)n u(n)

1
(1 z 1 )2

(n + 1)(n + 2) . . . (n + r 1) n
1
u(n)
(r 1)!
(1 z 1 )r

20.

21. n cos n u(n)

1 z 1 cos
1 z 1 2 cos + 2 z 2

22. n sin n u(n)

z 1 sin
1 z 1 2 cos + 2 z 2

23. Fn u(n)

(I z 1 F)1

B.3.2

Single-Sided Z-transform of non-causal signals

Notation
X + (z) =

n=0

X (z) = X + (z)

x(n)z n

Single-sided z-transform, x(n) not

necessarily causal.
For causal signals

Shifting x(n)yields:
i) one step shift
x(n 1) z 1 X + (z) + x(1)
x(n + 1) zX + (z) x(0) z
ii) n0 step shift (n0 0)
x(n n0 ) z n0 X + (z) + x(1)z n0 +1 +
+x(2)z n0 +2 + . . . + x(n0 )
x(n + n0 ) z n0 X + (z) x(0)z n0 x(1)z n0 1 . . . x(n0 1)z

51

B.4

Fourier Transform for Discrete Time Signal

1. X(f ) = F(x(n)) =

j2f 
=
= x()e

Transform
= 2f

 1/2

2. x(n) = 1/2 X(f )ej2f n df =

1 
jn
d
= 2
X(f )e
3.

a x (n)

Inverse transform

a X (f ) Linearity

4. x(n n0 )

X(f ) ej2f n0 Shift

5. x(n)ej2f0 n

X(f f0 ) Frequency translation

6. x(n) cos 2f0 n

1
[X(f f0 ) + X(f + f0 )]
2
Modulation

7. x(n) sin 2f0 n

1
[X(f f0 ) X(f + f0 )]
2j
Modulation

8. x y

X(f ) Y (f ) Convolution

9. x y

10.

= x()y() =
 1/2
= 1/2 X(f )Y (f )df

 1/2

1/2

X() Y (f )d Product

Parsevals teorem
for real time sequences

11. X(f ) = X (ej )

if x(n) = 0 for n < n0 and

2
= |x()| <
(Valid for for example: 18,19,20,21 och 22
in the Z-transform table for || < 1)


12. (n)

13. (n n0 )

ejn0

14. 1 n

15. u(n)

52

p=

(f p)

1 
1
1
+
p= (f p) +
2
2 j 2 tan(f )

16. 2f1 sinc(2f1 n) = 2f1



f
rectp 2f
1

sin(2f1 n)
2f1 n

|f n| < f1 < 1/2 , n integer

0 Else

Ideal LP lter
17. 4f1 sinc(2f1 n) cos(2f0 n)
rectp

f f0 + rect f + f0
p
2f1
2f1

Ideal BP Filter

18. 2f1 n cos 2f1 n2 sin 2f1 n

n

j2(f n) |f n| < f1 < 1/2 , n integer

(j2f )p =

0
Derivating system
19. cos(2f0 n)
20. |n|

Else

1 
[(f f0 p) + (f + f0 p)]
2 p=

1 2
1 + 2 cos 2f
2

21. |n| cos(2f0 n)

2
1
1
+
1 2
1 + 2 2 cos 2(f + f0 ) 1 + 2 2 cos 2(f f0 )

22.

M 1

1 |n|
2
M odd
pr (n) =

0 Else

Pr (f ) =

sin(f M )
Rectangular window
sin(f )

53

B.5

Some DFT Properties

Time
x(n), y(n)
x(n) = x(n + N )
x(N 1)
x((n 1))<N >
x(n)ej21n/N
x (n)
x1 (n) 
x2 (n)
x(n) 
y (n)
x1 (n)x2 (n)
N 1

n=0 x(n)y (n)

Frequency
X(k), Y (k)
X(k) = X(k + N )
X(N k)
X(k)ej2k1/N
X((k 1))<N >
X (N k)
X1 (k)X2 (k)
X(k)Y (k)
1
X (k) N X2 (k)
N 1
N 1
1

k=0 X(k)Y (k)

N

54