ECE411 - 5 - Frequency Analysis of DT Systems

The Fourier Transform and Frequency Analysis of
Continuous-Time Signals
Frequency analysis of a signal involves the resolution of
ECE 411 the signal into its frequency (sinusoidal) components.
The basic mathematical representation of periodic
Signals, Spectra, and signals is the Fourier series, which is a linear weighted
Signal Processing sum of harmonically related sinusoids or complex
exponentials.
Frequency Analysis of
Discrete-Time Systems
2 ECE 411 - Frequency Analysis of DT Signals and Systems
Frequency Analysis of Periodic DT Signals

The Discrete-Time Fourier Series (DTFS)
The Fourier series is denoted by Just as we have a representation of CT signals as a
linear weighted sum of complex exponentials, we can
x(t ) c
k
k cos(2 kF0t ) or x(t ) c e
k
k
j 2 kF0t
also apply the same to DT signals.
However, the frequency components of a DT signal are
where F0 = 1/Tp is the fundamental frequency, Tp is the not infinite unlike a CT signal.
fundamental period of the signal, kF0 are multiples or The frequency range (spectra) for DT signals are only
harmonics of the fundamental frequency where k is an unique over the interval to + or 0 to 2.
integer from to +, and ck are coefficients of the
Consequently, the DTFS will contain at most N
expansion.
frequency components separated by 2/N radians.
3 ECE 411 - Frequency Analysis of DT Signals and Systems 4 ECE 411 - Frequency Analysis of DT Signals and Systems
1
Frequency Analysis of Aperiodic DT Signals
DTFS of Periodic Signals The Discrete-Time Fourier Transform (DTFT)
Given a periodic sequence x(n) with period N, the DTFS of The Discrete-Time Fourier Transform (DTFT) of an aperiodic
discrete-time signal x(n) is defined as
x(n) consists of N harmonically-related exponentials.

x ( n)e

j n
X ( ) | x ( n ) |
N 1 N 1
N n
j 2 k 1
for:
x(n) ck e j 2 kfn or x(n) ck e n n
k 0 k 0 X() is a decomposition of x(n) into its frequency components, and

where: f = 1/N is a continuous variable.
X() is periodic from the range 0 to 2, thus any frequency outside
where ck are coefficients of the expansion. of this interval is equivalent to a frequency within the interval

X ( 2 k ) x(n)e
n
j ( 2 k ) n

x(n)e
n
j n j 2 k n
e X ()
Conversely, we can get x(n) from X(). Since is a continuous Assuming that XN() converges to X(), the function can be
variable, we integrate the function from to + after multiplying integrated to
both sides by ejm. 1
X ( )e
j n

x(n) d
X ( ) e j m
d x ( n ) e j n e j m d 2

n
The RHS of the function can be integrated if and only if
N
X N ( ) x(n)e
n N
j n
converges to X() as N .
This is due to the fact that x(n) is aperiodic, and no value of N
satisfies x(N + n) = x(n) for all values of n. XN() would converge to
X() only if we include as many samples as possible from x(n).
2
Relationship of the DTFT to the Z-Transform

Recall that X (z) x(n)z
n
n
ROC: r1 z r2
Since z = rej, therefore Frequency Analysis of

x(n)re x(n)r e

j n j n
X ( z) n
for r1 r r2 DT Signals and Systems
n n
*Note: X(z) can be interpreted as the DTFT of the signal x(n)rn
If r > 1, decaying Properties of the
r < 1, growing
Discrete-Time Fourier Transform (DTFT)
r = 1, X(z) = X()

x ( n )e j n Therefore, the DTFT is the Z-transform

X ( ) of the sequence evaluated on the unit
n
circle (r = 1).
Linearity Time Shifting

If x1(n)
F
X1() and x2 (n)
F
X2 () If x(n)
F
X ()
then x(n) a1x1(n) a2 x2 (n)

F
X () a1 X1() a2 X2 () then x(n k)
F
e j k X ()

Example: Determine the Fourier transform of the signal Proof: Z[x(n k)] zk x(n)zn
n
x(n) = a|n| where 1 < a < 1

k
z X(z) But z re j
e j |r 1
Therefore F[ x(n k )] e jk X ()
3
Time Reversal Convolution
If x(n)
F
X () If x1(n)
F
X1() and x2 (n)
F
X2 ()
then x(n)
F
X () then x(n) x1(n) x2 (n)

z
X () X1() X2 ()
Correlation Frequency Shifting

If x1(n)
F
X1() and x2 (n)
F
X2 () If x(n)
F
X ()

then rx1x2 (l) x1(n)x2 (n l)
F
SX1X2 () X1() X2 () then e j 0 n x(n)
F
X ( 0 )
n
Proof: Let x(n)' e j n x(n)

0

then F[ x(n)' ] x(n)' e j n
n
[e
n
j 0 n
x(n)]e j n

Therefore X ( 0 ) x(n)e
n
j ( 0 ) n
4
Modulation Theorem
If x(n)
F
X ()
x(n) cos(0n) 2 X ( 0 ) X ( 0 )
F
then 1
Frequency Analysis of
DT Signals and Systems
Proof: Since cos(0 n) 12 [e j n e j n ] 0 0
then x( n) cos(0 n) 12 [ x (n)e j n x (n)e j n ] 0 0

Analysis of DT-LTI Systems
in the Frequency Domain
But F [ x(n)e j n ] X ( 0 )
0
Therefore x(n) cos(0n) 12 X ( 0 ) X ( 0 )
Frequency Response of a DT-LTI System

Recall that y (n) h( k ) x ( n k )
k
Note that H ( ) h ( k )e j k
is a complex-valued function.
k
is the response of a relaxed DT-LTI system from an arbitrary input
signal x(n). To determine the frequency response of the system, we Therefore H ( ) h(k ) cos(k ) j h(k ) sin(k )
k k
excite the system with a complex exponential signal
x ( n ) Ae jwn H R ( ) jH I ( )

h(k )Ae

then y (n) j ( nk ) or y ( n ) A h ( k ) e j k e j n where: H R ( ) h(k ) cos( k ) Real part of H()
k k k

H I ( ) h( k ) sin( k ) Imaginary part of H()
where: k

Therefore y (n) AH ( )e j n
H ( ) h(k )e jk
DTFT of h(k)
Therefore: H ( ) H R ( ) H I ( )
2 2
( ) tan 1
H I ( )
k H R ( )
|h(n)| < for a stable system Magnitude response Phase response
5
Evaluation of the Frequency Response
of a DT-LTI System
M
It follows that the magnitude of H() is equal to
N M
[z z k ]
M
H ( z ) b0 z k 1
Since N V ( )
k
[z p ] k | H ( ) | b0 k 1
N
U
k 1
M k ( )
[ e j z k ] k 1
then H ( ) b0 (e j ) N M k 1
N and the phase of H() is equal to
[e p ]j
k
k 1 ( ) [ H ( )]
j k
Let: (e j
z k ) Vk ( )[e ] and (e j pk ) U k ( )[e j k ] M N

b0 ( N M ) k k
therefore Vk ( ) | e j z k | U k ( ) | e j pk | k 1 k 1
k ( ) e j z k
and
k ( ) e j pk
Recall that the evaluation of Fourier transform for the Observations:

given value of is equivalent to evaluating the z-transform The presence of a zero close to the unit circle causes
at point B on the unit circle. the magnitude of the frequency response, at
Using vector addition: frequencies that correspond to points of the unit circle
Im
OB OP PB and OB OZ ZB close to that point, to be small.
B
Note that: The presence of a pole close to the unit circle causes
OB e j |r 1 the magnitude of the frequency response to be large at
Z
OP pk OZ z k
P

frequencies close to that point.
O Re
Then
Placing a zero (or pole) close to the pole (or zero)
PB OB OP ZB OB OZ
j
cancels the effect of the pole (or zero)
e pk e j z k
j k ( )
U k ( )e Vk ( )e j k ( )
6
Example
>> b=[1 2 1];
>> a=[1 -0.9 0.81];
Given the system transfer function, determine the >> zplane(b,a)
magnitude and phase response at = /3. >>
1 2 z 1 z 2
H ( z)
1 0.9 z 1 0.81z 2
Answer: H ( ) 18.218 (25.210dB)
( ) 1.5404 (88.26)
= =0 = =0
Evaluate Location
H() at Location of poles
these
points
of zeros
7
>> b=[1 2 1];
>> a=[1 -0.9 0.81];
>> zplane(b,a) X: 0.3333
>> freqz(b,a,6) Y: 25.21
>>
X: 0.3333
Y: -88.26
LTI Systems as Frequency-Selective Filters Ideal Filter Characteristics

The term filter is a device that discriminates, according
|H()| |H()|
to some attribute, what passes through it.
1 B 1
In general, an LTI system modifies the input signal
spectrum X() according to its frequency response - -
-c 0 c -c 0 c
H() to yield an output spectrum Y().
Lowpass Highpass
H() acts as a weighting function or a spectral
shaping function to the different frequency components |H()| |H()|
in the input signal. 1 B 1
- -1 -c -2 0 1 c 2 - -c 0 c
Bandpass Bandstop
8
Ideal Filter Characteristics Ideal Filter Characteristics
We can design simple digital filters by placing poles Given the pole-zero placements, the system function
and zeros on the z-plane accordingly. can be expressed as
M M
Basically, we locate the poles near points on the unit b z k
k
N M
[1 z z k
1
] *Note: usually NM
circle to be emphasized, and place zeros near H ( z) k 0
N
b0 z k 1
N
to avoid any
1 ak z k
non-trivial zeros.
frequencies to be deemphasized.
k 1
[1 p z
k 1
k
1
]
Furthermore:
where b0 is a gain constant to normalize the
1. All poles should be placed inside the unit circle for the
frequency response at some specified frequency 0,
system to be stable. However, zeros can be placed anywhere
on the z-plane. that is
H (0 ) 1
2. All complex zeros and complex pole must occur in complex-
conjugate pairs in order for the filter coefficients to be real. where 0 is some frequency in the passband of the
filter.
Lowpass and Highpass Filters Lowpass and Highpass Filters

In the design of lowpass filters, poles are placed near |H()|
1
the unit circle at points corresponding to low
frequencies ( = 0). Zeros should be placed near or on 0.5
the unit circle at points corresponding to high r

0 -
frequencies ( = ). Magnitude Response
0
The reverse holds true for highpass filters the poles

|()|
at high frequencies, and the zeros at low frequencies.

1st-order LPF with

-
zero at the origin
-
1
H ( z ) b0 Phase Response
1 rz 1
9
To determine the value of b0, we set the magnitude |H()|
1
response to 1 at = 0.
1
H ( 0) 1 b0
0.5
1 r (e j0 ) 1 r
0 - 0
1
1 b0 Magnitude Response
1 r
|()|
b0 (1 r )

The transfer function becomes 1st-order LPF with a

-
zero at z = 1
1 -
H ( z ) (1 r ) 1 z 1
1 rz 1 H ( z ) b0 Phase Response
1 rz 1

We use the same procedure as in the first LPF to |H()| |H(c)|2 =
1
determine the value of b0.
1 (e j0 ) 1
H ( 0) 1 b0
0.5
r
1 r (e j0 ) 1 2 0
0 - -c 0 c
2 -0
1 b0 r Magnitude Response
1 r
(1 r ) |()|
b0
2
c
The transfer function becomes 2nd-order LPF with a
- -c
two zeros at z = 1
(1 r ) 1 z 1 (1 z 1 ) 2
-
H ( z) H ( z ) b0
1 rz 1
Phase Response
2 (1 pz 1 )(1 p z 1 )
10
For an LPF with complex-conjugate poles, we set the To determine the value of the poles, we use the cut-off
passband gain as H() = 1 to solve for b0. frequency gain.
At the passband: At the cut-off frequency:
j 0 2 2
(1 e ) 1 (1 e jc ) 2
H ( 0) 1 b0 2
H ( c ) b0
(1 pe j 0 )(1 p e j0 ) 2 (1 pe j c )(1 p e jc )
(1 1) 2 1 (1 e j c ) 2
1 b0 b0
(1 p (1))(1 p (1)) 2 (1 pe jc )(1 p e jc )
(1 p )(1 p )
b0 Substitute the value for b0 to solve for the poles.
4
(1 2r cos(0 ) r 2 )
b0
4

|H()| |H()|
1 1
0.5 0.5
r r
0 - 0 0 - 0
Magnitude Response Magnitude Response
|()| |()|

1st-order HPF with 1st-order HPF with a

- -
zero at the origin zero at z = 1
- -
1 1 z 1
H ( z ) b0 Phase Response H ( z ) b0 Phase Response
1 rz 1 1 rz 1
11
|H()| |H(c)|2 = The normalizing constant for HPFs can be solved using
1
the same procedures outlined for LPFs.
0.5
0
r 2
0 - -c 0 c
r
-0 Magnitude Response
|()|

-c
2nd-order HPF with a
- c
two zeros at z = 1
-
(1 z 1 ) 2
(1 pz 1 )(1 p z 1 )
Filter Transformations:
Example Lowpass-to-Highpass Transformations
Design a lowpass filter with the frequency response It is possible to change a low-pass filter to a high-pass
given as filter (and vice-versa) by applying the frequency
H (0) 1 translation property of the DTFT.
2 1 Say we are given a low-pass filter with the pole-zero
H ( 4 ) plot below:
2 The system transfer function
b0 b0 is therefore:
H ( z) 1 2
z 1
1 a1 z a2 z (1 pz 1 ) 2 H LP ( z )
r zr
0
0.46 And the frequency response

Answer: H ( z) e j 1
(1 0.32 z 1 ) 2 H LP ( )
e j r
12
Filter Transformations: Filter Transformations:
Lowpass-to-Highpass Transformations Lowpass-to-Highpass Transformations
|H()| By shifting the frequency by , we can transform the
low-pass response into a high-pass response.
1
The frequency response is now
e j ( ) 1
0.5
H LP ( )
e j ( ) r
e j e j 1
r
- 0
e j e j r
e j (1) 1
Magnitude Response 0
(Lowpass filter) e j (1) r
(e j 1)

( e j r )

|H()|
e j 1
H LP ( ) 0.5
e j r
Therefore
- 0
r
H LP ( ) F [(e j ) n hlp (n)]
0
Magnitude Response
H HP ( ) (Highpass filter)
z 1
H HP ( z )
zr
13
2 j k ( EVEN ) j
Mathematically: But e e 4 j e 2 kj e 1
We apply the frequency shifting property on the j 3 j k ( ODD ) j
And e e e ( 2 k 1) j e 1
systems frequency response
where k is an integer
1 b1e j b2 e 2 j bM e Mj
H LP ( )
1 a1e j a2 e 2 j a N e Nj Then
1 b1e j (1) b2 e 2 j (1) bM e Mj (1) M
1 b1e j ( ) b2e 2 j ( ) bM e Mj ( ) H HP ( )
H LP ( )
1 a1e j (1) a2 e 2 j (1) a N e Nj (1) N
1 a1e j ( ) a2 e 2 j ( ) a N e Nj ( )
1 b1e j b2 e 2 j b3e 3 j bM e Mj (1) M
1 b1e j e j b2e2 j e2 j bM eMj eMj H HP ( )
H HP () 1 a1e j a2 e 2 j a3e 3 j a N e Nj (1) N
1 a1e j e j a2e2 j e2 j aN e Nj e Nj
Sign changes for terms with odd exponents
Example Bandpass Filters

Given the system transfer function of a low-pass filter, The same principle applies for bandpass filters. A pair
find the transformed high-pass equivalent transfer of complex-conjugate poles are placed near points on
function and plot the 5-point frequency response. the unit circle that correspond to the passband of the
filter. Zeros are placed on either side of the poles.
1 2 z 1 z 2
H LP ( z )
1 0.9 z 1 0.81z 2
1 2 z 1 z 2
Answer: H HP ( z )
1 0.9 z 1 0.81z 2
14
Bandpass Filters Bandpass Filters
|H()| We solve for b0 by setting the magnitude at the
1
passband frequency 0 at 1.
r 0.5
1 e2 j0
0 H (0 ) 1 b0
0 - -1 -c -2 0 1 c 2 (1 re j0 e j0 )(1 re j0 e j0 )
-0 Magnitude Response 1 e2 j0
r 1 b0
|()|
(1 r)(1 re2 j0 )
2
-1 -c -2 1 e2 j0 b0 1 e2 j0
2nd-order BPF with H (0 ) 1 b0
- (1 r)(1 re2 j0 ) (1 r) 1 re2 j0
c = 0 and zeros at z = 1 1 c 2
-
1 z2 b0 (1 e2 j0 )(1 e2 j0 )
H (z) b0 Phase Response
(1 2r cos(0 )z 1 r 2 z2 ) (1 r) (1 re2 j0 )(1 re2 j0 )
Bandpass Filters Example
b0 1 e2 j0 e2 j0 1
Design a two-pole bandpass filter with the center
1 frequency at = /2 and frequency response given as
(1 r) (1 re2 j0 re2 j0 r 2 )
b0 2 2 cos(20 ) H (0) H ( ) 0
2 1
(1 r) 1 2r cos(20 ) r 2 H ( 49 )
2
b0 2 1 cos(20 ) 1 z 2
H ( z ) b0
(1 r) 1 2r cos(20 ) r 2 (1 2r cos(0 ) z 1 r 2 z 2 )
(1 r) 1 2r cos(20 ) r 2 1 z 2
b0 Answer: H ( z ) 0.15
2 1 cos(20 ) 1 0.7 z 2
15
Digital Resonators Digital Resonators
A digital resonator is a special two-pole bandpass filter |H()| r = 0.80
1 r = 0.95
with a pair of complex-conjugate poles located near the
unit circle. r 0.5
0
The filter has a large magnitude response (i.e. it
2
0 -
-r 0 r
resonates) in the vicinity of the pole location. -0 Magnitude Response
r
The angular position of the pole determines the r = 0.80
|()|
resonant frequency of the filter.
r = 0.95
-r
Applications of digital resonators include bandpass Resonator with
-
filtering and speech synthesis. r = 0 and zeros at the origin r
-
1
(1 2r cos(0 ) z 1 r 2 z 2 )

The output of the filter at the resonant frequency is 1
H ( ) 1 b0
2 j0
maximum. Solving for the normalization factor b0 (1 r ) (1 re )(1 re2 j0 )
1 1
H ( 0 ) 1 b0 1 b0
(1 2r cos(0 )e j0 r 2e2 j0 ) (1 r ) 1 re 2 j0
re2 j0 re2 j0 re2 j0
1
1 b0 1
1 (re j0
)e j0
1 (re j0
)e j0 1 b0
(1 r ) 1 r (e 2 j0
e 2 j0 ) r 2
1
1 b0 1
(1 r )(1 re2 j0 ) 1 b0
(1 r ) 1 2r cos(20 ) r 2
1
1 b0
(1 r ) (1 re2 j0 ) 2
b0 (1 r ) 1 2r cos(20 ) r 2
16
If zeros are placed at z = 1, the system function The output at the resonant frequency is maximum.
becomes Therefore
1 z 2 21 cos(20 )
H ( z ) b0 H (0 ) 1 b0
j (0 0 )
(1 2r cos(0 ) z 1 r 2 z 2 ) (1 re )(1 re j (0 0 ) )
21 cos(20 )
With the frequency response characteristic b0
(1 r )(1 re j 20 )
2 j
1 e
H ( ) b0 2 1 cos(20 )
(1 re j (0 )
)(1 re j (0 ) ) b0
(1 r ) (1 re j 20 ) 2
21 cos(2 ) 2 1 cos(20 )
H ( ) b0 b0
j (0 )
(1 re )(1 re j (0 ) ) (1 r ) (1 re j 20 )(1 re j 20 )

2 1 cos(20 ) |H()| r = 0.80
b0 1 r = 0.95
(1 r ) 1 2 cos(20 ) r 2
r 0.5
0
(1 r ) 1 2 cos(20 ) r 2
b0 0 - -c 0 c
2 1 cos(20 ) -0 Magnitude Response
r
Which is same as the normalization factor for the |()| r = 0.80

r = 0.95
bandpass filter. -c
Resonator with
-
r = 0 and zeros at z = 1 c
-
1 z 2
(1 2r cos(0 ) z 1 r 2 z 2 )
17
Notch Filters Notch Filters
A notch filter is a bandstop filter that contains one or |H()|
1
more deep notches (nulls) in its frequency response.
Notch filters are useful in applications where specific 0.5
0
frequency components need to be eliminated (e.g.
2
0 -
-c 0 c
60 Hz power line hum and it harmonics). -0 Magnitude Response
To create the null in the frequency response, a pair of
|()|
complex-conjugate zeros are placed on the unit circle
at some angle 0. -c
Notch filter with
-
zeros at 0 c
-
H ( z ) b0 (1 e j0 z 1 )(1 e j0 z 1 )
Phase Response
b0 (1 2 cos(0 ) z 1 z 2 )
Notch Filters Notch Filters

Since the magnitude response is pulled down near the Notch filters are usually implemented as FIR filters.
null, the value of b0 depends on the location (angle) of The problem with the FIR notch filter is that is has a
the null. wide bandwidth, severely attenuating frequencies
If 0 < 0 < /2, the magnitude at H( = 0) is low. Therefore around the null.
Poles can be introduced at the vicinity of the null to
H ( ) 1 b0 (1 2 cos(0 )e j e 2 j )
improve the frequency response, at the expense of
1 b0 (1 2 cos(0 )(1) 1) some resonance (ripple) near the null.
1 The IIR notch filter system function is
b0
2 2 cos(0 ) 1 2r cos(0 ) r 2
0 0 2 : b0
1 2 cos(0 ) z z
1 2
2 2 cos(0 )
For /2 < 0 < , H( = 0) = 1. Therefore H ( z ) b0
1 1 2r cos(0 ) z 1 r 2 z 2 ) 1 2r cos(0 ) r 2
b0

2 0 : b0
2 2 cos(0 ) 2 2 cos(0 )
18
Notch Filters Comb Filters
|H()| A comb filter is a special notch filter wherein the nulls
1
are periodically located along the frequency space.
r = 0.80
r 0.5
r = 0.95 This makes the frequency space look like a comb.
0
0 - -c 0 c Comb filters finds applications where rejection of a
-0 Magnitude Response certain frequency and its harmonics are needed.
r
|()|
Examples of its applications include power line
r = 0.80
r = 0.95 harmonics rejection, and in the suppression of clutter
-c
Notch filter with poles from fixed objects in moving-target indicator (MTI)
-
near zeros at 0 c radar systems.
-
1 2 cos(0 ) z 1 z 2
1 2r cos(0 ) z 1 r 2 z 2 )
Comb Filters Comb Filters

To derive the frequency response characteristic, With the frequency response as
consider a moving average FIR filter 1 [1 e j ( M 1) ]
M H ( )
1
y(n) x(n k ) M 1 (1 e j )
M 1 k 0
1 j M2 sin ( M21 )
e
The system function is M 1 sin( 2 )
M
1
H ( z) z k
We note that there are M zeros evenly spaced from
M 1 k 0
0 to 2 where
2
1 [1 z ( M 1) ] k k
M 1
M 1 (1 z 1 )
The pole at z = 1 is actually cancelled by the zero
at z = 1.
19
Comb Filters All-Pass Filters
|H()| An all-pass filter is defined as a system that has a
1
constant magnitude, that is
H ( ) 1 0
0.5
M 0
0 - 0
Magnitude Response The simplest form of an all-pass filter is a pure delay
system
|()|
H ( z) z k

Comb filter with M This system passes all signals without modification
-
zeros spaced 0 apart except delaying the samples by k units in time. This
M
- trivial system has a linear phase characteristic.
1
H ( z ) b0 z k Phase Response
M 1 k 0
All-Pass Filters All-Pass Filters

A more interesting all-pass system is defined by its 1
Let H1 ( z ) And H 2 ( z) BN ( z) AN ( z 1 )
denominator polynomial AN ( z)
N
AN ( z) ak z k , a0 1
k 0
Then H ( z) H1 ( z ) H 2 ( z) AN ( z ) AN ( z 1 ) 1
And its numerator polynomial The system is an all-pass system.
N 1 N N
BN ( z ) z AN ( z ) reverse polynomial of AN
Note that BN ( z ) a z k
N k
z 1
pk
H ( z) k 0
k 1
N N
The system function is therefore AN ( z )
1
a z k
k
1 p z k
1
BN ( z ) AN ( z ) k 0 k 1
H ( z) z N
AN ( z ) AN ( z ) Then if pk is a pole of the system, then 1/pk is a
zero of the system.
20
|H()|
The general form of the system is
1
NR
z 1 k NR
( z 1 k )( z 1 *k ) k real roots
H ( z) 0.5
1
k 1 1k z 1
k 1 (1 k z )(1 *k z ) k complex roots
1 1
r
r
0 - 0
For causal and stable systems Magnitude Response
1 k 1 and k 1
|()|

All-pass filters find applications as phase correction
(phase equalization) filters. When placed in cascade 1st-order all-pass filter
-
with a system with undesired phase characteristics,
-
the phase equalizer system is designed to z 1 1
compensate for the poor phase response. H ( z ) b0 r Phase Response
1 rz 1

1
r
|H()|
As with the other filters, b0 is computed at a
1
passband frequency (in this case, any frequency)
r
0.5 by setting it magnitude response to 1.

0
0 - 0 e r

1 e j0 e j 0 1 e j0
j0
r

-0 H ( 0) 1 b0
Magnitude Response

1 re j0 e j 0 1 re j0 e j 0
1 1r e 1 1r e
r
|()| j0 j0

1
b0
2nd-order all-pass filter
r
1 re 1 re
j0 j0
-
1 2 cos(0 ) 1 2

H ( z ) b0

z 1 e j0 z 1 1 e j0
1
r r
-
Phase Response
b0
r r
1 2r cos(0 ) r 2
j0 1
1 re z 1 re z
j0 1

21
All-Pass Filters
1 2 cos(0 ) 1 2
r r
1 b0
1 2r cos(0 ) r 2
1 2r cos(0 ) r 2
b0
1 2 cos(0 ) 1 2
r r
22

ECE411 - 5 - Frequency Analysis of DT Systems

Загружено:

Сведения о документе

Авторское право

Доступные форматы

Поделиться этим документом

Поделиться или встроить документ

Параметры публикации

Этот документ был вам полезен?

Это неприемлемый материал?

Авторское право:

Доступные форматы

ECE411 - 5 - Frequency Analysis of DT Systems

Загружено:

Авторское право:

Доступные форматы

The Fourier Transform and Frequency Analysis of

2 ECE 411 - Frequency Analysis of DT Signals and Systems

Frequency Analysis of Periodic DT Signals

k 0 k 0 X() is a decomposition of x(n) into its frequency components, and

Since z = rej, therefore Frequency Analysis of

x ( n )e j n Therefore, the DTFT is the Z-transform

Linearity Time Shifting

then x(n) a1x1(n) a2 x2 (n)

x(n) = a|n| where 1 < a < 1

Correlation Frequency Shifting

Proof: Let x(n)' e j n x(n)

then x( n) cos(0 n) 12 [ x (n)e j n x (n)e j n ] 0 0

Therefore x(n) cos(0n) 12 X ( 0 ) X ( 0 )

17 ECE 411 - Frequency Analysis of DT Signals and Systems

Frequency Response of a DT-LTI System

Recall that the evaluation of Fourier transform for the Observations:

Answer: H ( ) 18.218 (25.210dB)

25 ECE 411 - Frequency Analysis of DT Signals and Systems

LTI Systems as Frequency-Selective Filters Ideal Filter Characteristics

Lowpass and Highpass Filters Lowpass and Highpass Filters

the unit circle at points corresponding to high r

The reverse holds true for highpass filters the poles

1st-order LPF with

The transfer function becomes 1st-order LPF with a

Lowpass and Highpass Filters Lowpass and Highpass Filters

Lowpass and Highpass Filters Lowpass and Highpass Filters

1st-order HPF with 1st-order HPF with a

0.46 And the frequency response

Filter Transformations: Filter Transformations:

Example Bandpass Filters

Bandpass Filters Example

Digital Resonators Digital Resonators

Digital Resonators Digital Resonators

Which is same as the normalization factor for the |()| r = 0.80

Notch Filters Notch Filters

Comb Filters Comb Filters

All-Pass Filters All-Pass Filters

All-Pass Filters All-Pass Filters

85 ECE 411 - Frequency Analysis of DT Signals and Systems

Вам также может понравиться