6.02 Fall 2012 Lecture #13: - Frequency Response - Filters - Spectral Content

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6.

02 Fall 2012

Lecture #13

Frequency response
Filters
Spectral content

6.02 Fall 2012 Lecture 13 Slide #1


Sinusoidal Inputs and LTI Systems

h[n]

A very important property of LTI systems or channels:

If the input x[n] is a sinusoid of a given amplitude,


frequency and phase, the response will be a sinusoid at the
same frequency, although the amplitude and phase may be
altered. The change in amplitude and phase will, in
general, depend on the frequency of the input.

6.02 Fall 2012 Lecture 13 Slide #2


Complex Exponentials as

Eigenfunctions of LTI System

x[n]=ejn h[.] y[n]=H()ejn

Eigenfunction: Undergoes only scaling -- by the frequency


response H() in this case:

H () h[m]e jm
m

= h[m]cos(m) j h[m]sin(m)

m m

This is an infinite sum in general, but is well behaved if

h[.] is absolutely summable, i.e., if the system is stable.

We also call H() the discrete-time Fourier transform (DTFT)


of the time-domain function h[.] --- more on the DTFT later.
6.02 Fall 2012 Lecture 13 Slide #3
From Complex Exponentials to Sinusoids

cos(n)=(ejn+e-jn))/2

So response to a cosine input is:

Acos(0n+0) H() |H(0)|Acos(0n+0+<H(0))

(Recall that we only need vary in the interval [,].)

This gives rise to an easy experimental way to determine


the frequency response of an LTI system.
6.02 Fall 2012 Lecture 13 Slide #4
Loudspeaker Frequency Response

SPL versus Frequency


(Speaker Sensitivity = 85dB)

100

97

94

91

88
SPL (dB)

85

82

79

76 -3dB @ 56.5Hz -3dB @ 12.5k Hz

73

70
10 100 1,000 10,000 100,000

Frequency (Hz)

Image by MIT OpenCourseWare.


Spectral Content of Various Sounds

Human Voice
Cymbal Crash
Snare Drum
Bass Drum
Guitar
Bass Guitar
Synthesizer
Piano

13.75 Hz- 27.5 Hz- 55 Hz- 110 Hz- 220 Hz- 440 Hz- 880 Hz- 1,760 Hz- 3,520 Hz- 7,040 Hz- 14,080 Hz-
27.5 Hz 55 Hz 110 Hz 220 Hz 440 Hz 880 Hz 1,760 Hz 3,520 Hz 7,040 Hz 14,080 Hz 28,160 Hz

Image by MIT OpenCourseWare.

6.02 Fall 2012 Lecture 13 Slide #6


Connection between CT and DT

The continuous-time (CT) signal

x(t) = cos(t) = cos(2ft)

sampled every T seconds, i.e., at a sampling


frequency of fs = 1/T, gives rise to the discrete-time
(DT) signal

x[n] = x(nT) = cos(nT) = cos(n)

So =

and = corresponds to = /T or f = 1/(2T) = fs/2


6.02 Fall 2012 Lecture 13 Slide #7
Properties of H()

Repeats periodically on the frequency () axis, with period 2,


because the input ejn is the same for that differ by
integer multiples of 2. So only the interval in [-,] is of interest!

6.02 Fall 2012 Lecture 13 Slide #8


Properties of H()

Repeats periodically on the frequency () axis, with period 2,


because the input ejn is the same for that differ by
integer multiples of 2. So only the interval in [-,] is of interest!

= 0, i.e., ejn = 1, corresponds to a constant (or DC, which


stands for direct current, but now just means constant) input,
so H(0) is the DC gain of the system, i.e., gain for constant inputs.

H(0) = h[m] --- show this from the definition!

6.02 Fall 2012 Lecture 13 Slide #9


Properties of H()
Repeats periodically on the frequency () axis, with period 2,
because the input ejn is the same for that differ by
integer multiples of 2. So only the interval in [-,] is of interest!

= 0, i.e., ejn = 1, corresponds to a constant (or DC, which


stands for direct current, but now just means constant) input,
so H(0) is the DC gain of the system, i.e., gain for constant inputs.

H(0) = h[m] --- show this from the definition!

= or , i.e., Aejn=(-1)nA, corresponds to the


highest-frequency variation possible for a discrete-time
signal, so H()=H(-) is the high-frequency gain of the system.

H() = (-1)m h[m] --- show from definition!

6.02 Fall 2012 Lecture 13 Slide #10


Symmetry Properties of H()

H () h[m]e jm
m

= h[m]cos(m) j h[m]sin(m)

m m

= C() jS()

For real h[n]:


Real part of H() & magnitude are EVEN functions of .
Imaginary part & phase are ODD functions of .

For real and even h[n] = h[n], H() is purely real.

For real and odd h[n] = h[n], H() is purely imaginary.

6.02 Fall 2012 Lecture 13 Slide #11


Convolution in Time <--->

Multiplication in Frequency

x[n] h1[.] h2[.] y[n]

x[n] (h2*h1)[.] y[n]

In the frequency domain (i.e., thinking about input-to-output


frequency response):

x[n] H1() H2() y[n]

i.e., convolution in time


has become multiplication
H()=H2()H1() in frequency!
6.02 Fall 2012 Lecture 13 Slide #12
Example: Deconvolving Output of

Channel with Echo

x[n] y[n] z[n]


Channel, Receiver
h1[.] filter, h2[.]

Suppose channel is LTI with

1[n]=[n]+0.8[n-1]

H1() = ?? = 1
jm
h [m]e
m
= 1+ 0.8ej = 1 + 0.8cos() j0.8sin()
So:
|H1()| = [1.64 + 1.6cos()]1/2 EVEN function of ;

<H1() = arctan [(0.8sin()/[1 + 0.8cos()] ODD .


6.02 Fall 2012 Lecture 13 Slide #13
A Frequency-Domain view of Deconvolution

x[n] y[n] z[n]


Channel, Receiver
H1() filter, H2()

Noise w[n]
Given H1(), what should H2() be, to get z[n]=x[n]?

H2()=1/H1() Inverse filter

= (1/|H1()|). exp{j<H1()}

Inverse filter at receiver does very badly in the presence of noise

that adds to y[n]:


filter has high gain for noise precisely at frequencies where
channel gain|H1()| is low (and channel output is weak)!
6.02 Fall 2012 Lecture 13 Slide #14
A 10-cent Low-pass Filter

Suppose we wanted a low-pass filter with a cutoff frequency of /4?

x[n] H/4() H/2() H3/4() H() y[n]

6.02 Fall 2012 Lecture 13 Slide #15


To Get a Filter Section with a

Specified Zero-Pair in H()

Let h[0] = h[2] = 1, h[1] = , all other h[n] = 0

Then H() = 1 + e-j + e-j2 = e-j ( + 2cos())

So |H()| = | + 2cos()|, with zeros at


arccos(-/2)

6.02 Fall 2012 Lecture 13 Slide #16


The $4.99 version of a Low-pass Filter,

h[n] and H()

6.02 Fall 2012 Lecture 13 Slide #17


)
Determining h[n] from H(
H () = h[m]e jm

jn

Multiply both sides by e and integrate over a

(contiguous) 2 interval. Only one term survives!

H ()e jn d = h[m]e j(mn)


d

<2 > <2 > m

= 2 h[n]

1
h[n] =
2
H ()e jn d
<2 >

6.02 Fall 2012 Lecture 13 Slide #18


Design ideal lowpass filter with cutoff

frequency C and H()=1 in passband

1
h[n] =
2
H ()e jn
d
<2 >

C
1
=
2
1 e jn d
C

sin(C n)
= , n0
n DT sinc function
= C / , n=0 (extends to in time,
falls off only as 1/n))
6.02 Fall 2012 Lecture 13 Slide #19
Exercise: Frequency response of h[n-D]

Given an LTI system with unit sample response h[n]


and associated frequency response H(),

determine the frequency response HD() of an LTI


system whose unit sample response is

h D[n] = h[n-D].

Answer: HD() = exp{-jD}.()

so : |HD()| = |()| , i.e., magnitude unchanged

<HD() = -D + <() , i.e., linear phase term added

6.02 Fall 2012 Lecture 13 Slide #20


e.g.: Approximating an ideal lowpass filter

h[n] H[]

Not
causal

300 0 300  0 

Idea: shift h[n] right to get

causal LTI system.

Will the result still be a

6.02 Fall 2012 lowpass filter? Lecture 13 Slide #21


Causal approximation to ideal lowpass filter

hC[n]= h[n-300] |HC[]|

0 300 600 0
n

Determine <HC()

6.02 Fall 2012 Lecture 13 Slide #22


DT Fourier Transform (DTFT) for

Spectral Representation of General x[n]

If we can write

1
h[n] =
2
H ()e jn d where H () = h[n]e jn
<2 > Any contiguous
n
interval of length

then we can write 2

1
x[n] =
2
X()e jn
d where X() = x[n]e jn
<2 > n

This Fourier representation expresses x[n] as


a weighted combination of e jn for all in [,].

X()d is the spectral content of x[n]


6.02 Fall 2012 in the frequency interval [, + d ] Lecture 13 Slide #23
Useful Filters

6.02 Fall 2012 Lecture 13 Slide #24


Frequency Response of Channels

6.02 Fall 2012 Lecture 13 Slide #25


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6.02 Introduction to EECS II: Digital Communication Systems


Fall 2012

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