Lecture XII: Ideal filters

Maxim Raginsky

BME 171: Signals and Systems

Duke University

October 29, 2008

Maxim Raginsky Lecture XII: Ideal filters

This lecture

Plan for the lecture:

1 LTI systems with sinusoidal inputs
2 Analog filtering
frequency-domain description: passband, stopband
amplitude and phase response of ideal filters
time-domain description
3 Detailed example: ideal lowpass filter
4 Nonideal filters

Maxim Raginsky Lecture XII: Ideal filters

LTI systems with sinusoidal inputs
Consider an LTI system with impulse response h(t):

x(t) h(t) y(t)

LTI
Z ∞
y(t) = x(t) ⋆ h(t) = x(t − λ)h(λ)dλ
−∞

In the frequency domain we have

Y (ω) = H(ω)X(ω),

where X(ω) = F [x(t)], H(ω) = F [h(t)], Y (ω) = F [y(t)].

Consider a sinusoidal input of the form

x(t) = A cos(ω0 t + θ0 ),

where A is the amplitude, ω0 is the frequency, and θ0 is the phase.

Maxim Raginsky Lecture XII: Ideal filters
LTI systems with sinusoidal inputs

x(t) = A cos(ω0 t + θ0 )
πA e−jθ0 δ(ω + ω0 ) + ejθ0 δ(ω − ω0 )
 
X(ω) =

Y (ω) = H(ω)X(ω)
πA e−jθ0 H(ω)δ(ω + ω0 ) + ejθ0 H(ω)δ(ω − ω0 )
 
=
πA e−jθ0 H(−ω0 )δ(ω + ω0 ) + ejθ0 H(ω0 )δ(ω − ω0 )
 
=
h
= πA e−j(θ0 +∠H(ω0 )) |H(ω0 )|δ(ω + ω0 )
i
+ej(θ0 +∠H(ω0 )) |H(ω0 )|δ(ω − ω0 )

y(t) = F −1 [Y (ω)]
= A|H(ω0 )| cos(ω0 t + θ0 + ∠H(ω0 ))

Thus, the action of the LTI system with impulse response h(t) on a
sinusoid with amplitude A, frequency ω0 and phase θ0 is to transform the
amplitude as A → A|H(ω0 )| and the phase as θ0 → θ0 + ∠H(ω0 ).

Maxim Raginsky Lecture XII: Ideal filters

LTI systems with sinusoidal inputs
Many signals encountered in practice are finite sums of sinusoids:
N
X
x(t) = Ak cos(ωk t + θk )
k=1

The action of an LTI system with impulse response h(t) on such an input
is, by linearity, given by
N
X
y(t) = Ak |H(ωk )| cos(ωk t + θk + ∠H(ωk ))
k=1

Thus, an LTI system changes the amplitude ratios A

Al and the relative
k

phases θk − θl among the different frequency components

k, l = 1, . . . , N :
Ak Ak H(ωk )
→ ·
Al Al H(ωl )
θk − θl → θk − θl + ∠H(ωk ) − ∠H(ωl )

Maxim Raginsky Lecture XII: Ideal filters

Analog filtering
These considerations naturally lead us to the notion of filtering:
processing of signals in order to enhance certain frequency components
and to reject certain others. For example, if a signal consists of a
low-frequency information-bearing portion and a high-frequency noise
portion, we can employ a filter to reject the high frequencies and thus
remove the noise.

We will look at four kinds of filters:

1 low-pass filters pass all frequencies in the range |ω| ≤ B, for some
B > 0 and reject all others
2 high-pass filters pass all frequencies in the range |ω| ≥ B, for some
B > 0 and reject all others
3 bandpass filters pass all frequencies in the range B1 ≤ |ω| ≤ B2
for some B1 , B2 > 0 with B1 < B2 and reject all others
4 bandstop filters pass all frequencies in the range |ω| ≤ B1 and
|ω| ≥ B2 for some B1 , B2 > 0 with B1 < B2 and reject all others

Maxim Raginsky Lecture XII: Ideal filters

Analog filtering: frequency domain description
It is convenient to look at filters in the frequency domain. For each of
the four kinds of filters, we will specify the amplitude response |H(ω)|
and the phase response ∠H(ω).

We start with the amplitude response. For the four filters we have
defined above we have:
|HLP(ω)| |HHP(ω)|
1
1

-B 0 B ω -B 0 B ω
lowpass highpass

|HBP(ω)| |HBS(ω)|
1
1

-B2 -B1 0 B1 B2 ω -B2 -B1 0 B1 B2 ω

bandpass bandstop

Maxim Raginsky Lecture XII: Ideal filters

Some filtering terminology

Given a filter H(ω), the set of frequencies ω such that |H(ω)| > 0 is
called the passband of the filter; the set of frequencies ω such that
|H(ω)| = 0 is called the stopband of the filter.

filter passband stopband

lowpass |ω| ≤ B |ω| > B
highpass |ω| ≥ B |ω| < B
bandpass B1 ≤ |ω| ≤ B2 |ω| < B1 and |ω| > B2
bandstop |ω| ≤ B1 and |ω| ≥ B2 B1 < |ω| < B2

If the input to a filter is a sinusoid A cos(ω0 t + θ0 ), then the amplitude of

the output will be equal to:
A, if the frequency ω0 is in the passband of the filter
0, if the frequency ω0 is in the stopband of the filter

Maxim Raginsky Lecture XII: Ideal filters

Ideal filters

Next, we need to specify the phase response ∠H(ω) of the filter. We will
call a filter H(ω) ideal if

1, if ω is in the passband
|H(ω)| =
0, if ω is in the stopband

and 
−ωtd , if ω is in the passband
∠H(ω) =
0, if ω is in the stopband
where td > 0 is some constant.

The reason for calling such filters “ideal” will become clear shortly.

Maxim Raginsky Lecture XII: Ideal filters

Phase response of ideal filters

∠HLP(ω) ∠HHP(ω)

Btd Btd
-B -B
0 B ω 0 B ω
-Btd -Btd

lowpass highpass

∠HBP(ω) ∠HBS(ω)
B2td B2td
-B1 -B2 B1td -B1 -B2 B1td
0 B1 B2 ω 0 B1 B2 ω
-B1td -B1td
-B2td -B2td

bandpass bandstop

Maxim Raginsky Lecture XII: Ideal filters

Ideal filters with sinusoidal inputs
Let’s see what happens when we feed a sinusoidal signal

x(t) = A cos(ω0 t + θ0 )

into an ideal filter H(ω). We have already seen that the output will be

y(t) = A|H(ω0 )| cos(ω0 t + θ + ∠H(ω0 )).

Since |H(ω)| = 1 when ω is in the passband and 0 when ω is in the

stopband, while ∠H(ω) = −ωtd when ω is in the passband and 0
otherwise, we can further write

A cos(ω0 (t − td ) + θ0 ), if ω0 is in the passband
y(t) =
0, if ω0 is in the stopband

In other words, if the frequency of the sinusoid ω0 is in the passband of

the filter, then the output y(t) of the filter is a time-delayed version of
the input x(t):
y(t) = x(t − td ).

Maxim Raginsky Lecture XII: Ideal filters

Ideal filters with sinusoidal inputs

This explains why we use the term “ideal:” an ideal filter does not distort
the input signal, only delays it (provided the input frequency is in the
passband).

We can generalize these results to periodic signals that can be

represented by sums of sinusoids,
∞
X
x(t) = Ak cos(ωk t + θt ),
k=1

as well as to aperiodic signals that have a Fourier transform,

x(t) ↔ X(ω). In the latter case, it is convenient to visualize the action
of the filter in the frequency domain.

Maxim Raginsky Lecture XII: Ideal filters

Detailed example: ideal lowpass filter

Let us consider in detail the lowpass filter whose amplitude and phase
response are given by

|HLP (ω)| = p2B (ω) and ∠HLP (ω) = −ωtd p2B (ω),

where p2B (ω) is a rectangle of unit height and width 2B centered at

ω = 0.

We have
HLP (ω) = e−jωtd p2B (ω),
so that the impulse response has the form
 
  B B
hLP (t) = F −1 e−jωtd p2B (ω) = sinc (t − td )
π π

Note that the frequency response HLP (ω) is bandlimited, hence the
impulse response hLP (t) cannot be timelimited. This implies that an ideal
lowpass filter is acausal and therefore cannot be operated in real time.

Maxim Raginsky Lecture XII: Ideal filters

Nonideal filters
In fact, it can be shown that any ideal filter is necessarily acausal, and
therefore cannot be operated in real time. In practice, we have to resort
to causal approximations of ideal filters. For example, an ideal lowpass
filter can be approximated by an RC filter whose frequency response is
described by
1
|HRC (ω)| = p and ∠HRC (ω) = tan−1 (−ωRC)
(ωRC)2 + 1

-B 0 B ω

-B
0 B ω

Maxim Raginsky Lecture XII: Ideal filters