Laboratory 9 July 23, 2002, Release v3.

0 EECS 206

Laboratory 9

Filter Design, Modeling, and

the z-Plane

9.1 Introduction
So far, we’ve been considering filters as systems that we design and then apply to signals
to achieve a desired affect. However, filtering is also something that occurs everywhere,
without the intervention of a human filter designer. At sunset, the light of the sun is filtered
by the atmosphere, often yielding a spectacular array of colors. A concert hall filters the
sound of an orchestra before it reaches your ear, coloring the sound and adding pleasing
effects like reverberation. Even our own head, shoulders, and ears form a pair of filters that
allows us to localize sounds in space.
Quite often, we may wish to recreate these filtering effects so that we can study them
or apply them in different situations. One way to do this is to model these “natural” filters
using simple discrete-time filters. That is, if we can measure the response of a particular
system, we would often like to design a filter that has the same (or a similar) response.
One of the goals for this laboratory is to introduce the use of discrete-time filters as
models of real-world filters. In particular, we will examine how to apply a modeling approach
to understanding vowel signals. This in turn will suggest a way that we might improve the
performance of the vowel classifier we developed in Lab 8 using an automatic modeling
method.
Another goal of this lab is to present a method of filter design called pole-zero placement
design. Working with this method of filter design is extremely useful for building an intuition
of how the z-plane “works” with respect to the frequency domain that you are already
familiar with. The design interface that we use for this task should help you to develop a
graphical understanding of how poles and zeros affect the frequency response of a system.
We will use this design methodology both to design a traditional “goal-oriented” lowpass
filter, and to do some filter modeling.

9.1.1 “The Question”

• How can we design filters for certain purposes?

• How can we model vowel production using discrete-time filters?

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9.2 Background
9.2.1 Filters and the z-transform
Previously, we have presented the general time-domain input-output relationship for a causal
filter1 given by the convolution sum:
X X
y[n] = x[n] ∗ h[n] = h[k]x[n − k] = x[k]h[n − k] , (9.1)
k k

where x[n] is the input signal, y[n] is the output signal, and h[n] is the filter impulse response.
Using the z-transform techniques described in Chapter 7 of DSP First, we can also describe
the input/output relationship in the z-domain as

Y (z) = H(z)X(z) , (9.2)

where X(z) is the z-transform of x[n], which is the complex-valued function, defined on the
complex plane2 by
X
X(z) = x[n]z −n , (9.3)
n

where Y (z) is the z-transform of y[n], defined in a similar fashion, and where H(z) is the
system function of the filter, which is a complex-valued function defined on the complex
plane by one of the following equivalent definitions:

1. The system function is the z-transform of the filter impulse response h[n], i.e
X
H(z) = h[n]z −n . (9.4)
n

2. For X(z) and Y (z) as defined above, the system function is given by

Y (z)
H(z) = . (9.5)
X(z)

The system function has a very important relationship to the frequency response of a
system, H(ω̂). The system function evaluated at ej ω̂ is equal to the frequency response
evaluated at frequency ω̂. That is,

H(ω̂) = H(ej ω̂ ) . (9.6)

We can derive
P this result from equation 9.4. If we let z = ej ω̂ , then we know that H(z) =
j ω̂ −j ω̂n
H(e ) = n h[n]e . This is simply the definition of a system’s frequency given its
impulse response h[n].
1 Note that in this lab, we will only be concerned with causal filters.
2 The complex plane is simply the set of all complex numbers. The real part of the complex number is
indicated by the x-axis, while the imaginary part is indicated by the y-axis.

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9.2.2 FIR Filters and the z-transform

For a causal FIR filter, one can easily determine the system function using either of the
equivalent definitions given above. However, let us highlight the use of the second definition,
which will be useful in the next subsection where the first definition is difficult to apply.
In particular, for a causal FIR filter with coefficients {b0 , . . . , bM }, the general time-domain
input-output relationship for a causal FIR filter is given by the difference equation

y[n] = b0 x[n] + b1 x[n − 1] + b2 x[n − 2] + · · · + bM x[n − M ] . (9.7)

Taking the z-transform of both sides of this difference equation yields

Y (z) = b0 X(z) + b1 X(z)z −1 + b2 X(z)z −2 + · · · + bM X(z)z −M

= X(z)(b0 + b1 z −1 + b2 z −2 + b3 z −3 + · · · + bM z −M ) , (9.8)

where we have used the fact that the z-transform of x[n − no ] is X(z)z −no . Dividing both
sides of the above by X(z) gives the system function:
Y (z)
H(z) = = b0 + b1 z −1 + b2 z −2 + b3 z −3 + · · · + bM z −M . (9.9)
X(z)
Notice that H(z) is a polynomial of order M . We can factor the above complex-valued
polynomial as3

H(z) = K(1 − r1 z −1 )(1 − r2 z −1 )(1 − r3 z −1 ) · · · (1 − rM z −1 ) , (9.10)

where K is a real number called the gain, and {r1 , . . . , rM } are the M roots or zeros of
the polynomial, i.e. the values r such that H(r) = 0. We typically assume that the filter
coefficients bk are real. In this case, the zeros may be real or complex, and if one is complex,
then its complex conjugate is also a zero. That is, complex roots come in conjugate pairs.
The very important point to observe now from equation (9.10) is that the system function
H(z) of a causal FIR filter is completely determined by its gain and its zeros. Therefore,
we can think of {K, r1 , . . . , rM } as one more way to describe a filter4 . We will see that
when it comes to designing an FIR filter to have a certain desired frequency response, the
description of the filter in terms of its gain and its zeros is by far the most useful. In other
words, the best way to design a filter to have a desired frequency response (e.g., a low pass
filter) is to appropriately choose its gain and zeros. One may then find the system function
by multiplying out the terms of equation (9.10), and then picking off the filter coefficients
from the system function. For example, the number multiplying z −3 in the system function
is the filter coefficient b3 . The specific procedure will be described shortly.
The fact that we may design the frequency response of a causal FIR filter by choosing
its zeros5 stems from the following principle:
If a filter has a zero r located on the unit circle, i.e. |r| = 1, then H(∠r) = 0,
i.e. the frequency response has a null at frequency ∠r. Similarly, if a filter has
a zero r located close to the unit circle, i.e. |r| ≈ 1, then H(∠r) ≈ 0, i.e. the
frequency response has a dip at frequency ∠r. In either case, H(ω̂) ≈ 0, when
ω̂ ≈ ∠r.
3 The Fundamental Theorem of Algebra guarantees that H(z) factors in this way.
4 Previous ways of describing a filter have included the filter coefficients, the impulse response sequence,
the frequency response function, and the system function.
5 The gain does not affect the shape of the frequency response.

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The above fact follows from the property that if ω̂ = ∠r and |r| = 1, then e j ω̂ = r, and so
H(ω̂) = H(ej ω̂ ) = H(r) = 0 . (9.11)
A similar statement shows H(ω̂) ≈ 0 when |r| ≈ 1 and/or ω̂ ≈ ∠r.
From this fact, we see that we see that we can make a filter block a particular frequency,
i.e. create a null or a dip in the frequency response, simply by placing a zero on or near the
unit circle at an angle equal to the desired frequency6 . On the other hand, the frequency
response at frequencies corresponding to angles that are not close to these zeros will have
large magnitude. The filter will “pass” these frequencies. The specific procedure to design
such a filter is the following.
1. Choose frequencies ω̂1 , ..., ω̂L at which the frequency response should contain a null or
a dip.
2. Choose zeros ri = ρi ej ω̂i , i = 1, . . . , L, with ρi = 1 or ρi ≈ 1, depending upon whether
a null or a dip is desired at frequency ω̂. For each ω̂i 6= 0 choose also a zero rj that is
the complex conjugate of ri . Let M be the total number of zeros chosen.
3. Form the system function H(z) = K(1 − r1 z −1 ) × · · · × (1 − rM z −1 ), where K is a
gain that we also choose.
4. Cross multiply the factors of H(z) found in the previous step so as to express H(z)
as a polynomial whose terms are powers of z −1 .
5. Identify the FIR filter coefficients {b0 , . . . , bM }, which are simply the coefficients of
the polynomial found in the previous step, as shown in equation (9.9).

9.2.3 IIR filters and rational system functions

We now consider IIR filters. The general time-domain input-output relationship for a causal
IIR filter is given by the difference equation
y[n] = b0 x[n] +b1 x[n − 1] + b2 x[n − 2] + · · · + bM x[n − M ]
+a1 y[n − 1] + a2 y[n − 2] + · · · + aN y[n − N ] . (9.12)
Here, we have the usual FIR filter coefficients, bk , but we also have another set of coefficients
ak , which multiply past values of the filter’s output. We will call the bk ’s the feedforward
coefficients and the ak ’s the feedback coefficients 7 . If the ak ’s are zero, then this filter reduces
to a causal FIR filter.
As an example, consider the simple IIR filter with difference equation:
1
y[n] = x[n] + y[n − 1] (9.13)
2
What is the impulse response of this filter? If we assume8 that y[n] = 0 for n < 0, one can
straightforwardly show that the impulse response is
µ ¶n
1
h[n] = , n ≥ 0, (9.14)
2
6 Since non-real zeros must occur in conjugate pairs, we must also place a conjugate zero on the unit

circle, i.e. a zero whose angle is the negative of the first.

7 In some texts (and in Matlab), the feedback coefficients are defined as the negatives of the a coefficients
k
given here. Because of this, you should always be sure to check which convention is used.
8 This assumption is one of the “initial rest conditions” discussed in chapter 8 of DSP First.

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which is never zero for any positive n. (Note that the impulse response is generally not
so simple to compute; this is an unusual case where the impulse response can be obtained
by inspection.) Thus, by introducing feedback terms into our difference equation, we have
produced a filter with an infinite impulse response, i.e., an IIR filter.
In general, computing the system function by taking the z-transform of the resulting
infinite impulse may not be trivial because of the required infinite sum, and also because it
may be difficult to find the impulse response. However, we can use the fact that H(z) =
Y (z)/X(z) to determine the system function. To do this, we first collect the y[n] terms on
the left side of the equation and take the z-transform of the result.
y[n] − a1 y[n − 1] − · · · − aN y[n − N ] = b0 x[n] + b1 x[n − 1] + · · · + bM x[n − M ]
(9.15)
Y (z) − a1 Y (z)z −1 − · · · − aN Y (z)z −N = b0 X(z) + b1 X(z)z + · · · + bM X(z)z −M
−1

(9.16)
Y (z)(1 − a1 z −1 − · · · − aN z −N ) = X(z)(b0 + b1 z −1 + · · · + bM z −M ) (9.17)
Y (z) b0 + b1 z −1 + · · · + bM z −M
H(z) = = (9.18)
X(z) 1 − a1 z −1 − · · · − aN z −N
Equation (9.18) shows the general form of the system function of an IIR filters. Since it is
the ratio of two polynomials, it is called a rational function 9 .
Just as we could factor the polynomial in equation (9.9), we can do the same with
equation (9.18) to yield
(1 − r1 z −1 )(1 − r2 z −1 )(1 − r3 z −1 ) · · · (1 − rM z −1 )
H(z) = K . (9.19)
(1 − p1 z −1 )(1 − p2 z −2 )(1 − p3 z −1 ) · · · (1 − pN z −1 )
The roots of the polynomial in the numerator, {r1 , . . . , rM }, are again called the zeros of
the system function. The roots of the polynomial in the denominator, {p 1 , . . . , pN } are
called the poles of the system function. K is again a gain factor that determines the overall
amplitude of the system’s output. As before, the zeros are complex values where H(z) goes
to zero. The poles, on the other hand, are complex values where the denominator goes to
zero and thus the system function goes to infinity10 . Again, we typically assume that the
filter coefficients bk and ak are real, so both the poles and zeros of the system function must
be either purely real or must appear in complex conjugate pairs.
Just as we could completely characterize an FIR filter by its gain and its zeros, we can
completely characterize an IIR filter by its gain, its zeros, and its poles. As in the FIR
case, this is typically the most useful characterization when designing IIR filters. As before,
if the system function has zeros near the unit circle, then the filter magnitude frequency
response will be small at frequencies near the angles of these zeros. On the other hand, if
there are poles near the unit circle, then the magnitude frequency response will be be large
at frequencies near the angles of these poles. With FIR filters we could directly design filters
to have nulls or dips at desired frequencies. Now, with IIR filters, we can design peaks in
the frequency response, as well as nulls. The specific procedure is the following.
1. Choose frequencies ω̂1 , ..., ω̂L at which the frequency response should contain a null, a
dip, or a peak.
9 Thisis a generalization of the terminology that the ratio of two integers is called a rational number.
10 Technically,because of a division by zero, H(z) is undefined at the location of a pole. However, the
magnitude of the system function becomes very large in the neighborhood of a pole.

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2. Choose zeros ri = ρi ej ω̂i at those frequencies at which a null or a dip should occur,
with ρi = 1 or ρi ≈ 1, as desired. For each such ω̂i 6= 0, choose also a zero rj that is
the complex conjugate of ri . Let M be the total number of zeros chosen.

3. Choose poles pi = ρi ej ω̂i at those frequencies at which a peak should occur, with
ρi = 1 or ρi ≈ 1 as desired. For each such ω̂i 6= 0 choose also a pole pj that is the
complex conjugate of pi . Let N be the total number of poles chosen.
−1 −1
4. Form the system function H(z) = K (1−r 1z )×···×(1−rM z )
(1−p1 z −1 )×···×(1−pN z −1 ) , where K is a gain that
we also choose.

5. Cross multiply the factors of H(z) found in the previous step and express H(z) as the
ratio of two polynomials whose terms are powers of z −1 .

6. Identify the IIR filter coefficients {a0 , . . . , aN , b0 , . . . , bM }, which are simply the coef-
ficients of the polynomials found in the previous step, as shown in equation (9.18).

Poles and zeros at the origin and at infinity

Here, we have defined our system functions in terms of negative powers of z. This is
because our general forms for FIR and IIR filters are defined in terms of time delays, and
multiplication of the z-transform of some signal X(z) by z −1 is equivalent to a time delay of
one sample. However, there are may be “hidden” poles and zeros when we express a system
function in this matter.
Consider first the system function for our FIR filter given by (9.9). If we try to evaluate
this system function at z = 0, we will immediately find that we are dividing by zero. Thus,
there is actually a pole at the origin of this system function. To reveal such “hidden” poles
and zeros, we express the system function in terms of positive powers of z. To do so, we
M
multiply by zzM , which yields

b0 z M + b1 z M −1 + b2 z M −2 + b3 z M −3 + · · · + bM
H(z) = . (9.20)
zM
By the Fundamental Theorem of Algebra, we know that the numerator polynomial has M
roots, and thus the system has M zeros. However, the denominator, z M , has M roots as
well, all at z = 0. This means that our causal FIR system function has M poles at the
origin.
In some cases, like the previous example, we find extra poles at the origin. In other
cases, we find extra zeros at the origin. For example, the filter y[n] = y[n − 1] + x[n], has
H(z) = 1−z1 −1 = z−1z
, from which we see there is one zero at the origin. In still other cases
we find zeros at infinity11 . For example, the filter y[n] = x[n − 1], has H(z) = z −1 = z1 ,
from which we see that there is a zero at infinity. In still other cases, we find combinations
of the previous cases. We will call poles and zeros located at the origin or at infinitytrivial
poles and zeros because they do not affect the system’s magnitude frequency response 12 . In
this laboratory, we will primarily be concerned with nontrivial poles and zeros (those not
at the origin or at infinity).
11 A zero at infinity means that |H(z)| → 0 as |z| → ∞. It can be shown that a causal filter can never

have a pole at infinity.

12 Trivial poles and zeros do affect the phase (and thus the delay or time shift) of a system.

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0.8

0.6

0.4

Imaginary Part
0.2
2
0

−0.2

−0.4

−0.6

−0.8

−1
−1 −0.5 0 0.5 1
Real Part

Figure 9.1: A pole-zero plot of an IIR filter.

Note that there will always be the same number of poles and zeros in a linear time-
invariant system, including both trivial and nontrivial poles and zeros. The total number
equals the M or N , whichever is larger. Such facts are useful for checking to make sure that
you have accounted for all poles and zeros in a system.
Note also that if one chooses filter coefficients such that the numerator and denominator
contain an identical factor, i.e. if ri = pj for some i and j, then these factors “cancel” each
other, i.e. the filter is equivalent to a filter whose system function has neither factor.

Pole-zero plots
It is often very useful to graphically display the locations of a system’s poles and zeros. The
standard method for this is the pole-zero plot. Figure 9.1 shows an example of a pole-zero
plot. This is a two-dimensional plot of the z-plane that shows the unit circle, the real and
imaginary axes, and the position of the system’s poles and zeros. Zeros are typically marked
with an ‘o’, while poles are indicated with an ‘x’. Sometimes, a location has multiple poles
and zeros. In this case, a number is marked next to that location to indicate how many
poles or zeros exist there. Figure 9.1, for instance, shows four zeros (two conjugate pairs),
two “trivial” poles at the origin, and one other conjugate pair of poles. Recall that zeros
and poles near the unit circle can be expected to have a strong influence on the magnitude
frequency response of the filter.

9.2.4 Graphical interpretation of the system function

If we take the magnitude of H(z), we can think of |H(z)| as defining a (strictly positive)
surface over the z-plane for which the height of the surface is given as a function of the
complex number z. Figure 9.2 shows an example of just such a surface. This system
function has two zeros (which form a complex conjugate pair) and two poles at the origin.
Notice that the unit circle is outlined on the surface |H(z)|. The height of the surface at
z = ej ω̂ (i.e., on the unit circle) defines the magnitude of the frequency response, |H(ω̂)|,
which is shown to the right of the surface.

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1
(A)

Imaginary Part
(B)
2
0
5
4 −1
−1 0 1
3
|H(z)|

Real Part
2 3
(C)
1

|H(ω)|
2
1
−1 0 1
0
−1 0
1 Imag(z) 0 1 2 3
Real(z) Discrete radian frequency, ω

Figure 9.2: (A) The z-plane surface defined by the system function H(z) = (1 −
ejπ/4 z −1 )(1 − e−jπ/4 z −1 ). (B) The corresponding pole-zero plot. (C) The corresponding
magnitude frequency response.

1
(A)

Imaginary Part
(B)
2
0

4 −1
3 −1 0 1
|H(z)|

Real Part
2
1 2 (C)
|H(ω)|

1
−1 1
0
0
−1 0
1 Imag(z) 0 1 2 3
Real(z) Discrete radian frequency, ω

Figure 9.3: (A) The z-plane surface defined by the system function H(z) =
1
. (B) The corresponding pole-zero plot. (C) The corresponding
(1−0.8ejπ/2 z −1 )(1−0.8e−jπ/2 z −1 )
magnitude frequency response.

1
(A)
Imaginary Part

(B)
0

4
−1
3 −1 0 1
|H(z)|

Real Part
2
1 2 (C)
|H(ω)|

1
−1 1
0
0
−1 0
1 Imag(z) 0 1 2 3
Real(z) Discrete radian frequency, ω

Figure 9.4: (A) The z-plane surface for a complicated system function with four poles
and four zeros. (B) The corresponding pole-zero plot. (C) The corresponding magnitude
frequency response.

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On Figure 9.2, we can see two points where the surface |H(z)| goes to zero; these are the
zeros of the system function. Notice how the surface is “pulled down” in the vicinity of these
zeros, as though it has been “tacked to the ground” at the location of the zeros. Near the
system’s zeros, the magnitude frequency response has a low point because of the influence
of the nearby zero. Also notice how the surface is “pushed up” at points far from the zeros;
this is another common characteristic of system function zeros. (Since the two poles in this
figure are at the origin, they have no effect on the system’s magnitude frequency response.)
Thus, the magnitude frequency response has higher gain at points far away from the zeros.
Figure 9.3 shows the surface |H(z)| as defined by a different system function. This
system function has two poles (which form a complex conjugate pair) and two zeros at the
origin. Notice how the poles “push up” the surface near them, like poles under a tent. The
surface then typically “drapes” down away from the poles, getting lower at points further
from them. The magnitude frequency response here has a point of high gain in the vicinity
of the poles. (Again, the zeros in this system function are located at the origin, and thus
do not affect the magnitude frequency response.)
Figure 9.4 shows the surface for a system function which has poles and zeros interacting
on the surface. This system function has four poles and four zeros. Notice the tendency
of the poles and zeros to cancel the effects of one another. If a pole and a zero coincide
exactly, they will completely cancel. If, however, a pole and a zero are very near one another
but do not have exactly the same position, the z-plane surface must decrease in height from
infinity to zero quite rapidly. This behavior allows the design of filters with rapid transitions
between high gain and low gain.

9.2.5 Poles and stability

System poles cause the system function to go to infinity at certain values of z because we
are dividing by zero. On the one hand, this can have the desirable effect of raising the
magnitude frequency response at certain frequencies. On the the other hand, this can have
some undesirable side effects. One somewhat significant problem is introduced if we have a
pole outside the unit circle. Consider the following filter, for instance:

y[n] = x[n] + 2y[n − 1] (9.21)

This filter has a single pole at z = 2. What is this system’s impulse response? If the input
is x[n] = δ[n] and y[n] = 0 for n < 0, then at n = 0, y[n] = 1. Then, every y[n] after that
is equal to twice the value of y[n − 1]. The value of this impulse response grows as time
goes on! This system is unstable 13 . Unstable filters cause severe problems, and so we wish
to avoid them at all costs. As a general rule of thumb, you can keep your filters from being
unstable by keeping their poles strictly inside the unit circle. Note that the system’s zeros
do not need to be inside the unit to maintain stability.

9.2.6 Filter design using manual pole-zero placement

In this laboratory, we will explore a method of filter design in which we place poles and
zeros on the z-plane in order to match some target frequency response. You will be using
a Matlab graphical user interface to do this. The interface allows you to place, delete,
13 Technically, it is bounded input, bounded output (BIBO) unstable because the input has a limited

magnitude but the output does not.

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and move poles and zeros around the z-plane. The frequency response will be displayed
in another figure and will change dynamically as you move poles and zeros. To keep the
filter’s coefficients real, you will design by placing a pair of poles and zeros on the z-plane
simultaneously.
The approach for this method depends somewhat on the type of filter that we wish to
design. If we want an FIR filter (i.e., a filter that has no poles), we need to use zeros to
“pin down” the frequency response where it is low, and allow the frequency response to be
pushed upwards in regions where there are no zeros. Note that if we put a zero right on
the unit circle, we introduce null in the frequency response at that point. Conversely, the
closer to the origin that we place a zero, the less effect it will have on the frequency response
(since it will begin to affect all points on the unit circle roughly equally). You might use
the example of the running average filter and bandpass filters (given in Chapter 7 of DSP
First) as a prototype of how to use zeros to design FIR filters using zero placement.
If we wish to design an IIR filter (with both poles and zeros), it usually makes sense to
start with the poles since they typically affect the frequency response to a greater extent.
If the frequency response that we are trying to match has peaks on it, this suggests that we
should place a pole somewhere near that peak (inside the unit circle). Then, use zeros to
try to pull down the frequency response where it is too high. As with zeros, poles near the
origin have relatively little effect on the system’s filter response.
Regardless of which type of filter we are designing, there are a couple of methodological
points that should be mentioned. First, moving a pole or zero affects the frequency response
of the entire system. This means that we cannot simply optimize the position of each pole-
pair and zero-pair individually and expect to have a system which is optimized overall.
Instead, after adjusting the position of any pole-pair or zero-pair, we generally need to move
many of the remaining pairs to compensate for the changes. This means that filter design
using manual pole-zero placement is fundamentally an iterative design process.
Additionally, it is important that you consider the filter’s gain. Often we cannot adjust
the overall magnitude of the frequency response using just poles and zeros. Thus, to match
the frequency response properly, you may need to adjust the filter’s gain up or down. The
pole-zero design interface that you will use in this Lab includes an edit box where you can
change the gain parameter. Alternately, by dragging the frequency response curve, you can
change the gain graphically. A related idea is that of spectral slope. By having a pair of
poles or zeros inside the unit circle and near the real axis, we can adjust the overall “tilt”
of the frequency response. As we move the pair to the right and left on the z-plane, we can
adjust the slope of the system’s frequency response up and down.
Note that there are automatic filter design methods which do not require manual place-
ment of poles and zeros. In Section 9.2.8 we discuss one such method.

9.2.7 Design of Standard Filter Types

Many of the filters that we wish to design and use belong to one of four standard types:
lowpass, highpass, bandpass, or bandstop. These filters are characterized by a passband (a
band of frequencies which are relatively unaltered by the filter) and a stopband (a band of
frequencies which are significantly attenuated, or decreased in amplitude, by the filter). The
locations of the passband and stopband are what characterize the different filter types. For
instance, a lowpass filter has a passband which contains low frequencies and a stopband
which contains high frequencies, while a bandpass filter has a single passband that is sur-
rounded by two stopband regions. Between the passband and the stopband is a transition

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Transition
Band
Passband Stopband

Passband Stopband
Ripple Attenuation
|H(ω)|

Frequency Cutoff Frequency

Figure 9.5: An illustration of the various bands of a lowpass filter.

band in which the filter’s frequency response changes from high to low. The location of the
transition band in frequency determines the filter’s cutoff frequency. In this lab, we will
specify the cutoff frequency by using the two frequencies that bound the transition band.
When designing these types of filter, there are a number of different design goals that
we may attempt to achieve. For instance, we may wish to have a very flat frequency
response in the passband of the filter. Unfortunately, it can be difficult to achieve a flat
frequency response over some frequency region. Instead, the frequency response usually
varies somewhat over that region; this variation is called ripple. Thus, one of design criteria
may be to minimize passband ripple. In this lab, we define the passband ripple as the
(positive) decibel value of the ratio between the maximum and minimum filter gains in the
passband.
Another common goal is to try to minimize the gain in the stopband of the filter relative
to the gain in the passband. That is, we wish to maximize the stopband attenuation. In this
lab, we define the stopband attenuation as the decibel ratio between the maximum filter
gain in the passband and the maximum filter gain in the stopband.
Figure 9.5 shows the passband, transition band, and stopband for a lowpass filter. The
figure also illustrates the the passband ripple and stopband attenuation. In Problem 2 of
this assignment, you will design a lowpass filter to maximize stopband attenuation.

9.2.8 Modeling Vowel Production

In Lab 8, we discussed some of the properties of vowel production in speech, but we did not
examine the mechanisms behind vowel production. In order to gain a better understanding
of vowel production and how we can model it using discrete-time filters, we need to introduce
some theory.
Speech production is primarily governed by the larynx (or voice box) and the vocal tract.
Figure 9.6 shows a diagram of the larynx and vocal tract. When we speak a vowel, the lungs
push a stream of air through the larynx and the vocal folds (commonly referred to as the
vocal chords). Given the appropriate muscular tension, this stream of air causes the vocal

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Figure 9.6: A diagram showing the larynx and vocal tract.

x[n]
Glottal Vocal Tract y[n]
Source Filter

Figure 9.7: A block diagram of the source-filter model of speech production.

folds to vibrate14 . This in turn creates a nearly periodic fluctuation in air pressure passing
through the larynx. The fundamental frequency of vocal fold vibration is typically around
100 Hz for males and 200 Hz for females.
This fluctuating air stream then passes through the vocal tract, which is the airway
leading from the larynx and through the mouth to the lips. The positions of the tongue,
lips, and jaw serve to shape the vocal tract, with different positions creating different vowel
sounds. The different sounds are produced as the vocal tract shapes the spectrum of the
pressure signal coming from the larynx. Depending upon the vocal tract configuration,
different frequencies of the spectrum are emphasized; from Lab 8, we know these frequencies
as formants. When whispering a vowel, the lungs push air through the larynx, but the vocal
folds do not vibrate. In this case, the air pressure fluctuation is quite noise-like and generally
is not periodic. Nevertheless, the tongue, lips and jaw shape the vocal tract just as before
to make the various vowel sounds.
Note that the above description is only accurate for vowels and so-called voiced conso-
nants like “m” and “n.” Most consonant sounds are produced using the tongue, lips, and
teeth rather than the vocal cords. We will not consider consonants in this lab.
It is traditional to model speech production using a source-filter model. Figure 9.7 shows
a block diagram of the source-filter model. The first block is the glottal source, which
14 In fact, during normal production the vocal folds open and close completely on each cycle of the vibration.

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X(ω)
0.5

0
0 0.5 1 1.5 2 2.5 3

1
H(ω)

0.5

0
0 0.5 1 1.5 2 2.5 3

1
Y(ω)

0.5

0
0 0.5 1 1.5 2 2.5 3
Frequency (radians per sample)

Figure 9.8: A plot of the magnitude spectrum of a glottal source signal, the frequency
response of a vocal tract filter, and the magnitude spectrum of the output signal.

takes as input a fundamental frequency and produces a periodic signal (the glottal source
signal ) with the given fundamental frequency. The signal produced is typically modeled as
a periodic pulse train. To a first approximation, we can assume that spectrum of this pulse
train is composed of equal amplitude harmonics. The glottal source signal is meant to be
analogous to the signal formed by the air pressure fluctuations produced by the vibrating
vocal cords. Note that to model whispering, the glottal source signal can be modeled using
random noise rather than a pulse train.
The second block of the source-filter model is the vocal tract filter. This is a discrete-
time filter that mimics the spectrum-shaping properties of the vocal tract. Since we are
assuming a source signal with equal-amplitude harmonics, the vocal tract filter provides
the spectral envelope for our output signal. That is, when we filter the source signal with
fundamental frequency ω̂0 radians per sample, the k th harmonic of the output signal will
have an amplitude equal to the filter’s magnitude frequency response evaluated at k ω̂ 0 . This
is illustrated in Figure 9.8 which shows a particular example. The magnitude spectrum of
the glottal source signal is shown on top, the magnitude frequency response of the vocal
tract filter is shown in the center, and the magnitude spectrum of the output signal, which
is the signal that models the specific vowel signal. One may clearly see that, as desired, the
envelope of the spectrum of the vowel signal model matches the spectrum of the vocal tract
filter.
In Problem 3 of this assignment, we will make such source-filter models for particular
vowel signals, by measuring the spectrum of the vowel signal and designing an IIR vocal
tract filter whose frequency response approximates this spectrum.
Typically, our vocal tract filter can have relatively few filter coefficients (i.e., approxi-
mately 10-20 coefficients). Further, the acoustics of the vocal tract suggest that this filter
should be IIR. Often, the vocal tract is modeled using an all-pole filter which has no non-

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trivial zeros. This is because an acoustic passageway like the vocal tract primarily affects
a sound through resonances. A resonance is a part of a system that tends to vibrate at a
certain resonant frequency, thus amplifying that frequency in signals passed through them.
The feedback form of an IIR filter is a direct implementation of resonance; this is how IIR
filters are able to produce high gain at certain frequencies. Using this simple model of speech
production, it is possible to synthesize artificial vowels.

All-pole analysis and vowel classification

In the last lab, we explored some features for vowel classification that were based on two
measures of spectral energy in a vowel signal. The development of the source-filter model,
however, suggests an acoustically motivated feature for vowel classification. If we assume
that the vocal tract can be modeled with a low-order discrete-time filter, then the vocal
tract filter captures all of the relevant information about which vowel has been produced.
Variations such as fundamental frequency and type of vowel production (i.e., voiced or
whispered) are restricted to the glottal source and can be neglected. Using samples of the
frequency response of the vocal tract filter as features for vowel classification has been shown
to produce good classification results.
Fortunately, there are nice mathematical tools for deriving all-pole filter models auto-
matically from a time-domain waveform. These tools, fit the spectrum of a time-domain
signal with poles in a least-squares sense. Note that these tools work directly with the time-
domain waveform rather than its spectrum; typically, they return the resulting a k feedback
coefficients for a filter with those poles, rather than the locations of the poles themselves.
We will explore these tools for all-pole analysis in the laboratory assignment, and we will
compare classification performance using features based on these models to the performance
we achieved with our other feature sets from Lab 8.

9.3 Some Matlab commands for this lab

• Calculating the frequency response of IIR Filters: Previously we have used
freqz to compute the frequency response of FIR filters. We can use the same command
to compute the frequency response of an IIR filter. If our filter is defined by feedforward
coefficients bk stored in a vector B and feedback coefficients ak stored in a vector A,
we compute the frequency response at 256 points using the command:

>> [H,w] = freqz(B,A,256);

As with FIR filters, Matlab’s convention for the bk coefficients is B(1)= b0 , B(2)= b1 ,
. . ., B(M+1)= bM . Matlab’s convention for the ak coefficients is A(1)= 1, A(2)= −a1 ,
. . ., A(N+1)= −aN . Both bk and ak are given as defined in equation 9.18. H contains
the frequency response and w contains the corresponding discrete-time frequencies. Al-
ternatively, we can compute the frequency response only at a desired set of frequencies.
For example, the command

>> [H,w] = freqz(B,A,[pi/4, pi/2, 3*pi/4]);

returns the frequency response of the filter at the frequencies π/4, π/2, and 3π/4.

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1
0.8
0.6
0.4
0.2
Imag(z) 2
0
−0.2
−0.4
−0.6
−0.8
−1

−1 −0.5 0 0.5 1
Real(z)

2
3

|H(z)|
2
1.5
|H(ω)|

1 1
0 0.5
−1 0
−0.5
0 −0.5
0.5 0.5
1 −1
0 0.5 1 1.5 2 2.5 3 Imag(z)
ω (Discrete radian frequency) Real(z)

Figure 9.9: The GUI window for Pole-Zero Place 3-D.

• Pole-Zero Place 3-D: In this laboratory, we will primarily be exploring filter design
using manual pole-zero placement. To help us do this, we will be using a Matlab
graphical user interface (GUI) called Pole-Zero Place 3-D. Pole-Zero Place 3-D allows
you to place, move, and delete poles and zeros on the z-plane, and provides immediate
feedback by displaying the filter’s frequency response and the |H(z)| surface. Addi-
tionally, it calculates some useful statistics for assessing the quality of a particular
filter design.
To run this program you need to download two different files: pole_zero_place3d.m
and pole_zero_place3d.fig. To begin Pole-Zero Place 3-D, simply execute 15 the
command

>> pole_zero_place3d;

Once the program starts, the GUI window shown in Figure 9.9 will appear. The axis
in the upper left of the window shows a portion of the z-plane with the unit circle.
In the lower left is an axis that displays the frequency response of the system. In the
lower right is a 3-D axis which displays a 3-D graph of the |H(z)| surface16 .
The interface allows you to do a wide variety of things.

1. To add a poles or zeros to the z-plane, click the Add Zeros or Add Poles button
and then click on the z-plane plot in the upper left of the GUI. The state of the
Place pair checkbox determines whether a single (real) pole or zero is added, or
whether a conjugate pair is added.
15 This program was designed to run using Windows systems running Matlab 6 or higher; it will not work

with previous versions of Matlab. It should work with Unix operating systems running Matlab 6, but this
has not been tested.
16 This surface plot requires significant computation, and thus it can be toggled on and off using the View

3-D checkbox in the upper right.

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Note that the program also adds the hidden poles and zeros that accompany
nontrivial poles and zeros. Specifically, for each zero that is added a pole is
added at the origin, or if there is already at least one zero at the origin, instead
of adding a pole at the origin, one zero at the origin is removed, i.e. cancelled.
Moreover, for each pole that is added, a zero is added at the origin, or if there
are already poles at the origin, one pole is removed, i.e. cancelled. The system
does not allow one to place zeros at infinity, and it can be shown that zeros at
infinity will not be induced by any other choices of poles or zeros.
2. To move a real pole or zero (or a conjugate pair of complex poles or zeros), you
must first select the pole/zero by clicking on one member of the pair. Then, you
can drag it around the z-plane, use the arrow keys to move it, or move it to
a particular location by inputting the magnitude and angle (in radians) in the
Magnitude and Angle edit boxes.
3. To delete a pole or zero (or pair), select it and hit the Delete Poles/Zeros button.
Again, the system will maintain an equal number of poles and zeros by also
removing poles or zeros from the origin as necessary. This may also have the
effect of no longer cancelling other poles and zeros, and thus the total number of
poles and zeros that appear at the origin will change.
4. To change the filter’s gain, you can either use the Filter Gain edit box or you
can click-and-drag the blue frequency response curve in the lower left.
5. To toggle between linear amplitude and decibel displays in the lower two plots,
select the desired radio button above the Filter Gain edit box.
6. To rotate the 3-D |H(z)| plot, simply click-and-drag the axes in the lower right
of the GUI. To enable or disable the 3-D plot, toggle the View 3-D checkbox in
the upper right of the GUI
7. To begin with an initial filter configuration defined by the feedforward coefficients,
B, and the feedback coefficients, A, start the program with the command
>> pole_zero_place3d(B,A);
This is useful if you wish to start continue working on a design that you had
previously saved. You may set either of these parameters to empty ([]) if you
do not wish to specify the filter coefficients.
8. To print the GUI window, you can either use the Copy to Clipboard button to
copy an image of the figure into the clipboard17 , or you can print the figure using
the Print GUI button.
9. To save your current design, use the Export Filter Coefs button. The feedfor-
ward and feedback coefficients will be stored in the variables B_pz and A_pz,
respectively.
10. To hear your filter’s response to periodic signal with equal-amplitude harmonics,
press the Play Sound button. This is particularly useful when using the GUI to
design vocal tract filters for vowel synthesis.

• Pole-Zero Place 3-D – Filter Matching Mode: In “filter matching mode,” you
specify samples of a desired transfer function at harmonically related frequencies and
17 Windows operating systems only

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try to match that transfer function. The GUI plots a red curve or stem plot along
with the frequency response function; this is the response we wish to match. Two
edit boxes labeled Linear Matching Error and Decibel Matching Error indicate how
closely your filter matches the desired frequency response. The matching error values
are computed as the RMS error between the desired frequency response and your filter
design in both linear amplitude and in decibels.
To start the GUI in this mode, use the following command:

>> pole_zero_place3d(B,A,filter_gains,fund_frq);

filter_gains are the values of the desired filter frequency response at harmonically
related frequencies, and fund_frq is the fundamental frequency (in radians per sam-
ple) of the harmonic series at which filter_gains are defined.

• Pole-Zero Place 3-D – Lowpass Design Mode: In “lowpass design mode,” you
specify the maximum frequency of the passband and the minimum frequency of the
stopband (both in radians per sample). To start the GUI in this mode, use the
following command:

>> pole_zero_place3d(B,A,[pass_max,stop_min]);

In this mode, the GUI computes the passband ripple and the stopband attenuation of
your lowpass filter design. You can use these measures to evaluate your filter design.
The figure in the lower right also displays the passband and stopband of the filter,
with appropriate minima and maxima.

• Converting between filter coefficients and zeros-poles: Given a set of filter

coefficients, we often need to determine the set of poles and zeros defined by those
coefficients. Similarly, we often need to take a set of poles and zeros and compute the
corresponding filter coefficients. There are two Matlab commands that help us do
this. First, if we have our filter coefficients stored in the vectors B and A, we compute
the poles and zeros using the commands

>> zeros = roots(B);

>> poles = roots(A);

This is because the system zeros are simply the roots of the numerator polynomial
whose coefficients are the numbers in B, while the system zeros are simply the roots of
the denominator polynomial whose coefficients are the numbers in A. We can see this
in equation 9.18. To convert back, use the commands

>> B = poly(zeros);
>> A = poly(poles);

Note that we loose the filter’s gain coefficient, K, with both of these conversions.

• Generating pole-zero plots: Frequently, we’d like to use Matlab to make a pole-
zero plot for a filter. If our filter is defined by feedforward coefficients B and feedback
coefficients A (both row vectors), we can generate a pole-zero plot using the command:

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>> zplane(B,A);

Alternately, if we have a list of poles, p, and a list of zeros, z, (both column vectors)
we can use the following command:

>> zplane(z,p);

An example of a pole-zero plot resulting from this command is shown in Figure 9.1.

• Automatic all-pole modeling: Using the Matlab command aryule, we can com-
pute an all-pole filter model for a discrete-time signal. That is aryule automatically
finds an all-pole filter whose magnitude frequency response that in some sense matches
the magnitude frequency response of the signal. If signal is given by signal, the com-
mand

>> A = aryule(signal,N);

returns the filter feedback coefficients ak as a vector A. The parameter N indicates

how many poles we wish to use in our filter model. Once we have A, we can compute
the filter’s frequency response at 256 points using freqz as

>> [H,w] = freqz(1,A,256);

9.4 Demonstrations in the Lab Section

• The z-transform, system functions, and IIR filters

• The system function “surface,” |H(z)|

• Using Pole-Zero Place 3-D for filter design

• Vocal tract modeling

9.5 Laboratory Assignment

1. (Fit an FIR filter’s frequency response.) Download the files pole_zero_place3d.m,
pole_zero_place3d.fig, and lab9_data.mat. In this problem, you will get familiar
with the Pole-Zero Place 3-D program for filter design using pole-zero placement.
In lab9_data, the variable FIR_fr contains samples of the frequency response for a
simple FIR filter with six zeros. Execute Pole-Zero Place 3-D using the command

>> pole_zero_place3d([],[],FIR_fr,2*pi/8192);

Use the GUI to find an FIR filter with six nontrivial zeros that matches the frequency
response of the original filter. You should be able to get the linear matching error to
be less than 0.1. (Hint: The original filter had all six of its zeros inside the unit circle,
so yours should as well.)

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• Include the GUI window with your matching filter in your report. In this and
the following problems, make sure that it is possible to read the filter evaluation
scores on your printout. (Note: The easiest way to include the GUI window
is to use the Copy to Clipboard button on a Windows machine. After hitting
the button, wait for the GUI to flash white and then paste the result into your
report.)
• What are the filter coefficients bk and ak for your filter?
• Where are the zeros on the z-plane? Give your answers in rectangular form.

2. (Design a lowpass filter.) In this problem, we will use the “lowpass design mode” of
Pole-Zero Place 3-D to design some lowpass filters, as described in Section 9.2.7. For
the various parts of this problem, use the command

>> pole_zero_place3d([],[],2*pi*[1500 2000]/8192);

This sets the filter transition band to 1500 Hz to 2000 Hz if we assume a sampling
rate of 8192 samples per second.

(a) (Design an FIR lowpass filter to maximize stopband attenuation.) First, let’s
see what we can do with just zeros (that is, with FIR filters). Using only six
nontrivial zeros (i.e., three zero pairs), design a lowpass filter with a stopband
attenuation of at least 30 dB. (Remember, we want our stopband attenuation to
be as large as possible). For now, take note of your filter’s passband ripple, but
don’t worry about minimizing it.
• Include the GUI window with your matching filter in your report.
• What are the filter coefficients bk and ak for your filter?
• Where are the zeros on the z-plane? Give your answers in rectangular form.
Food for thought: Using just zeros, try to find a way to minimize the passband
ripple. What does this do to your stopband attenuation? Try this with more zeros,
but don’t use any poles.
(b) (Design an IIR lowpass filter to maximize stopband attenuation.) There are two
primary benefits to the use of IIR filters. First, it is very easy to get very high
gain at certain frequencies. This lets us design a lowpass filter with very high
stopband attenuation. Using a single pair of nontrivial poles, design a lowpass
filter that has a stopband attenuation greater than 60 dB. Use the same transition
band as in the previous problem. Again, you should take note of the passband
ripple, but don’t worry about minimizing it.
• Include the GUI window with your matching filter in your report.
• What are the filter coefficients bk and ak for your filter?
• Where are the poles on the z-plane? Give your answers in rectangular form.
(c) (Design an IIR lowpass filter for both high stobpand attenuation and low ripple.)
The second benefit of IIR filters is the ability to achieve fast transitions between
high gain and low gain. Among other things, this allows us to transition between
the passband and stopband more quickly, which in turn allows us to achieve
relatively high stopband attenuation with low passband ripple.

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Once again using the same transition band, design a lowpass filter with a passband
ripple of less than 2 dB and a stopband attenuation of at least 20dB. You may
use as many poles and zeros as you wish, but it is possible to meet these criteria
with only two poles and four zeros. (Hint: use decibel mode to help you increase
the stopband attenuation, and linear mode to help you decrease the passband
ripple.)
• Include the GUI window with your matching filter in your report.
• What are the filter coefficients bk and ak for your filter?
• Where are the poles and zeros on the z-plane? Give your answers in rectan-
gular form.
Food for thought: For a more interesting challenge, design a lowpass filter with a
passband ripple of less than 1 dB and a stopband attenuation of 60 dB. This can
be done with six poles and six zeros, but you might want to use more than this.

3. (Matching a vowel signal’s spectrum using pole-zero placement.) lab9_data.mat con-

tains the variable vowel2, which is a short vowel signal sampled at 8192 Hz. In this
problem we will find a filter model of the vocal tract used to produce this vowel.

(a) First, let’s examine the signal itself.

• Use soundsc to listen to vowel2. What vowel does this signal represent?
• As a measure of the spectrum, plot the magnitude DFT coefficients for this
signal in decibels versus frequency in Hertz. Use only the first half of the
DFT coefficients, where the maximum value on the x-axis is one half of the
sampling rate.
• From this plot, estimate the fundamental frequency of the vowel signal in
Hertz.
(b) (Design a vocal tract filter with both poles and zeros.) lab9_data.mat contains
the variables vowel2_amps, which contains the amplitudes of the harmonics that
make up vowel2. We’ll use the amplitudes to find a vocal tract filter for this
vowel. Start Pole-Zero Place 3-D using the command
>> pole_zero_place3d([],[],vowel2_amps,2*pi*frq/8192);
where frq is the fundamental frequency (in Hz) that you estimated. (Hint: The
red stems should go all the way across the frequency response plot. Also, there
should be one stem for each element of vowel2_amps. If not, you have probably
estimated your fundamental frequency incorrectly.)
Set the GUI to “decibel plot” and find a filter with six nontrivial poles and six
nontrivial zeros that makes the decibel matching error as small as possible. You
should be able to get the decibel error below 4.2. (It is possible to achieve a
decibel error of 3.65.)
• Include the GUI window with your matching filter in your report.
• What are the filter coefficients bk and ak for your filter?
• Where are the poles and zeros on the z-plane? Give your answers in rectan-
gular form.

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(c) (Design a vocal tract filter with only poles and zeros.) Now, repeat the above for
a filter with 10 poles nontrivial and no nontrivial zeros (except for those at the
origin). You should be able to achieve a decibel matching error below 2.6. (It is
possible to achieve a decibel error of 2.1.)
• Include the GUI window with your matching filter in your report.
• What are the filter coefficients bk and ak for your filter?
• Where are the poles on the z-plane? Give your answers in rectangular form.
If you are working on a computer with audio capability, you should use the Play
Sound button to listen to a synthesis of the vowel signal. Note that the all-pole
model produces less error with fewer total coefficients. This suggests that all-pole
filters are more appropriate for vocal tract modeling.

4. (All-pole modeling and classification) In this problem, we’ll look at an automatically

generated all-pole model of vowel2, and we’ll see how we can use this model to help
us improve our vowel classifier performance from Lab 8.

(a) (Automatically generate an all-pole filter model.) Use aryule to compute an

all-pole model for vowel2 with 10 poles.
• What are the resulting feedback coefficients?
• Make a pole-zero plot of the resulting filter.
• Use freqz to plot the frequency response of this filter and the frequency re-
sponse of the filter you found in Problem 3c in two subplots of the same figure.
Display the two frequency responses in decibels. (Note: The two frequency
responses in decibels may be offset by some constant, which corresponds to
a scaling of the original spectrum.)
(b) (Construct and evaluate the vowel classifier.) Here, we’ll consider 16 samples
of the frequency response of an all-pole filter to be a potential feature vector for
vowel classification. lab9_data.mat contains five matrices which contain all-pole
feature vectors for the same vowel instances we examined in Lab 8. They are
oo_ap, oh_ap, ah_ap, ae_ap, and ee_ap.
• As you did in Lab 8, compute the mean all-pole feature vectors for each of
the five vowel classes. Make sure you label which mean vector belongs with
which class.
• Combine the above vectors into a matrix of mean feature vectors. Then,
use confusion_matrix.m (from Lab 8) to calculate the confusion matrix
for the all-pole features. As you did in Lab 8, use the vowel order “oo,”
“oh,” “ah,” “ae,” and “ee”. (Note: if you were unsuccessful at completing
confusion_matrix in Lab 8, you can use the compiled function confusion_matrix_demo.dll
for this problem.)
• Compare this confusion matrix with the two confusion matrices that you
computed in Lab 8. Is the performance of the classifier better with this
feature class? Note any similarities between the three confusion matrices.

5. On the front page of your report, please provide an estimate of the average amount of
time spent outside of lab by each member of the group.

252 The University of Michigan, All rights reserved