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4 ANALYSIS OF DT-SYSTEMS IN THE TIME-DOMAIN

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The most frequent task in systems analysis:

given: a SISO system (represented e.g. by its system equation), and an input signal x[nT].
to be determined: the systems response (its output signal) y[nT].
Variety of methods:
solving difference equations Mathematics.
recursive numerical method chapter 4.1
convolution chapter 4.2 + 4.3
z-transform chapter 5

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This method assumes that the system is causal so that its output at a given point in time only depends on present or past signal
samples. It requires that the system equation is available.

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4.1 The Recursive Numerical Method (C11.2)

A general representation for system equations of causal LTI systems is:

y[nT]

i 1

i 0

(-ai ) y[(n - i)T]  (bi ) x[(n - i)T]

From this representation the following scheme may be set up and used to subsequently determine the output samples of y[nT]:

b0
n
x[nT]
n0
...
n0+1 ...
...
...

b1
x[(n-1)T]
...
...
...

b2
x[(n-2)T]
...
...
...

...
...
...
...
...

input
bl
x[(n-l)T]
...
...
...

output
-ak
y[(n-k)T]
...(*)
...
...

...
...
... (*)
...
...

-a2
y[(n-2)T]
... (*)
...
...

-a1
y[(n-1)T]
... (*)
...
...

y[nT]
...
...
...

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Each row is used to calculate y[nT] from the other entries in the row and the coefficients in the header row according to:
y[nT] b0 x[nT]  b1 x[(n - 1)T]  ...  bl x[(n - l)T] - ak y[(n - k)T] - ... - a1 y[(n - 1)T]
For the algorithm to work it is required to
a) determine a starting point n0; this is usually the left border of the input signal and may in many cases be set to n0=0.
b) know about the initial conditions. Initial conditions are the samples that occurred at the output signal before the set starting point.
They are y[(n0-1)T], y[(n0-2)T], y[(n0-k)T], marked in the scheme by (*). Thus a system of order k requires k initial conditions. In
many cases all initial conditions are 0.
Such algorithm is well structured for hardware implementation, requiring multiply-add units and shift operations.
A drawback of this algorithm is that it only produces numerical values. A mathematical expression for y[nT] is often hard to determine
from them.

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4.2 Unit Pulse Response (C11.3)

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The unit pulse response of a system is the output signal when the input signal is a unit pulse function G[nT]; its symbol is h[nT].
It is unique for a given system and contains all information available on the system itself!
The unit pulse response of a system may be determined
with mathematical methods, or
potentially by applying the recursive numerical method, or
by use of the z-transform ( chapter 5; preferred method), or
estimated through measurements of a pulse or step response.
A system is causal if and only if h[nT]=0 for all n<0.

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4.3 Superposition Concept, Convolution, and some Properties (C11.4+5)

Given
an LTI system, uniquely specified by its unit pulse response h[nT], and
an input signal x[nT].
It can be shown that the output signal y[nT] can be calculated through the following sum:
f

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y[nT]

x[mT] h[(n - m)T] ;

m f

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Such operation is called convolution, with a short-form notation: y[nT]=x[nT] h[nT] .

Characteristics of the convolution operation:

convolution is commutative:
convolution is associative:
convolution is distributive (over addition):
if x[nT] and h[nT] show finite extent, left and right border usually add
during convolution, and the extent calculates from nye = nxe + nhe - 1.
the unit pulse function is the neutral element of convolution:
convoluting a signal with a shifted unit pulse simply shifts the signal:

x[nT] h[nT] = h[nT] x[nT]

(x[nT] h[nT]) g[nT] = x[nT] (h[nT] g[nT])
x[nT] (h[nT] + g[nT]) = x[nT] h[nT] + x[nT] g[nT]
x[nT] G[nT] = x[nT]
x[nT] G[(t-t0)T] = x[(t-t0)T]

The following is a detailed step-by-step approach to solve this convolution sum and to determine the system response in the timedomain; to ease the procedure it is usually performed considering only the index values and omitting the time axis. Note that with
increased experience one will not need to perform each individual step.
1) Sketch a graph of the input signal x[n] and the system impulse response h[n].
Write down a functional expression for x[n] and h[n].
2) Sketch a graph of x[m], h[m] and h[-m]; these are based on m as independent variable.
Sketch a graph of h[n-m] and denote its significant points.
Write down a functional expression for x[m] and h[n-m].

3) For each value of m in the range of -f..f (usually interval-wise):

Sketch a graph of the relationship between x[m] and h[n-m] over m.
Multiply x[m] and h[n-m] for each point on the m-axis.
Build the sum over that product (eventually piecewise): y[n]

x[m] h[n - m] .

m f

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4) Finally combine the partial results into a piecewise specification for y[n].
Generally the partial sums to be determined in step 3) are infinite sums and it may be difficult to specify a resulting expression; only
approximations might be available in such case.

Special cases:
n

x[m] h[n - m]

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if the system is causal: y[n]

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if input signal x[nT] and unit pulse response h[nT] both are finite with extents nxe, nhe, convolution can be performed in (nxe+nhe-1)
single steps, each one producing a single sample value for y[nT], and the remaining samples of y[nT] being 0.

m f

It may be worth noticing that the sum of the sample indices of x[m] and h[n-m] always is n.

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5 THE Z-TRANSFORM (C14)

Determining the output of an LTI system through the convolution operation may soon become tedious if the system's complexity
increases. Moreover, several important questions cannot be answered at all in the time-domain.
This chapter introduces a new concept for signal and systems analysis purposes, with support for some system design decisions: the
z- transform. It allows to determine LTI system responses in a systematic way (and more generally to ease the solution of linear
difference equations), and it also provides a quantitative measure for system stability.
A prerequisite for all considerations in this chapter (except parts of section 5.4) is that signals are zero for all times nT<0.This can be
achieved in many cases by just defining nT=0 at the very beginning of the processes under investigation.

5.1 Definition of the z-Transform

Given a signal x[nT] which can be interpreted as a mathematical function of the integer variable n, another mathematical function X(z)
of a complex variable z can be determined in many cases through following integral:
f

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X(z)

x[nT] z-n
n 0

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This is the so-called (single-sided) z-transform of x[nT].

Note that the lower limit of the sum could also have been set to -f, as x[nT] must be 0 for negative times.
The variable z is defined as z r e j2 Sf T . The ROC (radius of convergence) is the smallest radius rc of a circle around 0 so that the
sum exists for all r>rc.
One of the most important characteristics of the z-transform is that it allows to uniquely reconstruct x[nT] from X(z) using the inverse ztransform. Thus, transform and inverse transform can be applied subsequently and always switch between the same two functions.
1
n-1
nt0
j2S X(z) z dz
ri

The inverse z-transform is defined as x(nT)

, with ri>ROC, though it is usually not evaluated directly.
for

n0
0

Notations:
As z is often called a frequency in a wider sense, especially in the context of z-transforms, X(z) is then consequently called a signal
in the frequency-domain; also the term z-domain is widely used.
The operation to produce X(z) from x[nT] is called z-transform and is sometimes denoted by X(z)=Z
Z{x[nT]}; the operation to produce
x[nT] from X(z) is called inverse z-transform and is sometimes denoted by x[nT]=Z-1{X(z)}.
As the transform can be performed in both directions, a frequent notation is x[nT]
X(z) with the open circle representing the
time-domain and the filled circle representing the frequency-domain.
It has become common practice to specify signals in the time-domain with lowercase letters and signals in the frequency-domain
with uppercase letters.

Determining z-transform and inverse transform results:

There is a three-tier decision process to perform the transform of a time-domain signal x[nT] into X(z):
1. Determine whether the signal x[nT] can be found in a table of z-transform pairs.
2. Determine whether any transform theorems can be applied to the signal x[nT] so that the results can be found in a table of ztransform pairs.
3. If neither of the first two actions leads to the desired transform, the definition sum for X(z) must be evaluated.
The inverse transform of X(z) into its time-domain signal x[nT] can be achieved by following decision hierarchy:
1. Determine whether the signal X(z) can be found in a table of z- transform pairs.
2. Determine whether any transform theorems can be applied to the signal X(z) so that the results can be found in a table of ztransform pairs.
3. Determine whether partial fraction expansion can be applied to X(z) so that the results can be found in a table of z-transform pairs
or transform theorems can be applied.
4. If neither of the first three actions leads to the desired transform, the definition integral for the inverse transform must be evaluated.
Note, that for the inverse z-transform partial fraction expansion helps to avoid solving the integral in most cases.

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5.2 Theorems of the z-Transform

The following list holds a set of important theorems; they can be seen as rules helping to determine the z-transform of a signal.
R1: Linearity
a.X(z) + b.Y(z)
for a,b: const.
Let x[nT]
X(z) and y[nT]
Y(z)

a.x[nT]+b.y[nT]
R2: Time shift
Let x[nT]
X(z)

x[(n-m)T]
X(z).z-m
for mt0, m: const.
R3: z-scaling
z
Let x[nT]
X(z)

for a: const.
an x[nT]
X( )
a
k
R4: Multiplying with (nT)
d
Let x[nT]
X(z)

(nT)k x[nT]
 z T Z (nT)k 1 x[nT] for k>0, k: const.
dz
R5: Difference
Let x[nT]
X(z)

x[nT]-x[(n-1)T]
(1-z-1).X(z)
R6: Sum
n
X(z)
Let x[nT]
X(z)

x[iT]
1 - z-1
i 0
R7: Convolution
Let x[nT]
X(z) and y[nT]
Y(z)

x[nT] y[nT]
X(z) . Y(z)
R8: Limit Theorems
R9a: Initial Value
x[0]
lim (X(z))
Let x[nT]
X(z)

z of

R9b: Final Value

Let x[nT]
X(z)

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(x[f] ) lim x[nT]

lim (1 - z-1) X(z)

nof

z o1

only in case all poles of (1-z-1).X(z) lie inside the unit circle

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5.3 Inverse z-Transform

In many cases the z-transform X(z) of a signal x[nT] turns out to be a rational function of z, consisting of a numerator polynomial and a
denominator polynomial. If x[nT]=0 for n<0, a generalized representation can be given as follows:
b0  b1z-1  b2z-2  ...  br -1z-(r -1)  br z-r
; r,m; a0
X(z)
a0  a1z-1  a2z- 2  ...  am-1z-(m-1)  amz-m
where some of the b0..br-1 and a1..am-1 may be zero.
For real signals x[nT] the coefficients ai and bi in numerator and denominator polynomials will be real values, which makes X(z)
becoming a member of the very important mathematical class of rational functions with real coefficients.
In case only a finite number of samples of x[nT] are needed, power series expansion may be applied to X(z); it just produces a
sequence of sample values.
Otherwise partial fraction expansion must be performed, which finally leads to a mathematical expression for x[nT].
Both approaches are specified in the following sections.

5.3.1 power series expansion

X(z) can be expanded into a power series of (z-1) centered at 0. This can simply be achieved by a polynomial division.
X(z) (b0  b1z-1  b2z-2  ...  br -1z-(r -1)  br z-r ) y (a0  a1z-1  a2z-2  ...  am-1z-(m-1)  amz-m )
and results in a (usually infinitely long) series
s
X(z) c 0  c1z-1  c 2z-2  c 3 z-3  ...  c s z-s  ... ;
When stopping at z-s (i.e. cutting the series), the resulting term Xa(z) evolved so far approximates X(z) and can easily be transformed
into the time domain using the linearity and time shift theorems. The corresponding time signal xa[nT] matches the true x[nT] exactly
for ns and misses the trace of x[nT] completely for n>s.

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5.3.2 partial fraction expansion

Rational functions with real coefficients as given above can be expanded to only contain non-negative exponents of z and then
factorized in numerator and denominator, leading to their factor form:
(z - E1) (z - E2 ) ...
X(z) C zm-r
(z - D1) (z - D 2 ) ... (z - Dm )
This representation shows the following important characteristics:
The values Di are called the poles of X(z); the number m of poles of X(z) equals the order of the denominator polynomial. If m<r, a
further set of (r-m) poles appear at z=0.
Some of the poles may have identical values. If a value D appears k times, the pole s=D is said to have an order of k.
A pole may be a complex value. In this case a complex conjugate pole exists as well, and both can be combined to form a real
valued quadratic term in z.
The values Ei are called the zeros of X(z). They are usually not of importance in this context and not further investigated except for
the case when a zero has the same value as a pole; in that case their corresponding factors in numerator and denominator can be
reduced.

If a signal X(z) is a rational function as given above, it can be represented by a sum of simpler rational functions. This procedure is
called partial fraction expansion and is specified in Mathematics; thereby complex conjugate poles should be combined into a
quadratic term to make the further processes more comfortable.
Once X(z) is represented as a sum of its partial fractions there is a good chance to find a corresponding entry in the table of ztransform pairs for each of the remaining summands.
Before applying partial fraction expansion it must be verified that the polynomial order of the numerator is smaller than the polynomial
order of the denominator. If this is not the case, a polynomial division has to take place first, and partial fraction expansion is applied
only to the remaining rational function.
However it can be shown under the preconditions of this chapter for the expression X(z)/z that its numerator order is always smaller
than its denominator order, and therefore usually X(z)/z is expanded into partial fractions, thereby avoiding a polynomial division.
Afterwards each partial sum is multiplied with z (or divided by z-1) to achieve a representation for X(z).

Thus determining x[nT] from X(z) through partial fraction expansion is performed in several steps:
1. Write down X(z)/z as a polynomial in numerator and denominator.
2. Determine all poles of X(z)/z and express the denominator using product terms (z -Di); combine all complex conjugate pole pairs
into a quadratic term each.
3. Apply partial fraction expansion on X(z)/z.
4. Multiply the result with z (or, preferably, divide it by z-1), resulting in a sum of partial fractions for X(z).
5. Find the inverse z-transform for each partial fraction from the table of transform pairs, potentially also using transform theorems.
This results in an expression for x[nT].

5.4 Z-Transform of Linear Difference Equations

Systems are described by system equations.
System equations of LTI systems are linear difference equations. Given a causal LTI system with x[nT] as input and y[nT] as output
signal, a general form of the system equation is:
k

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y[nT]  ai y[(n - i)T]

i 1

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bi x[(n - i)T]
i 0

For real world systems all coefficients ai and bi are real values, as they originate from the difference equation above.
Such equations can undergo the z-transform as a whole, using the theorems above. Solving the resulting equation for Y(z) leads to:
b0  b1z-1  b2z-2  ...  bl-1z-(l-1)  blz-l
X(z)
Y(z)
a0  a1z-1  a2z- 2  ...  ak -1z-(k -1)  ak z-k
This is a good starting point to determine the response of a system on a given input signal x[nT]. The procedure to follow is:
1. Given a system, described by its system equation, build the z-transform of the system equation and solve it for Y(z).
2. Determine the z-transform X(z) of the given input signal x[nT].
3. Put X(z) into the result of step 1 and transform the expression for Y(z) into a rational function.
4. Follow the procedure in the previous section to obtain y[nT] from Y(z).
In case the system (and hence the system equation) does not change, step 1 needs to be executed only once.

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A very important aspect of this method is that it also can be applied in cases where the system output y[nT] has not been 0 for all
times nT<0, an exception from the precondition set up above for this whole chapter.
In such cases step 1 of the procedure must use the following modified time shift theorem for transforming the left hand side of the
system equation (here specified for y[nT]):
Let y[nT]

Y(z)

y[(n - m)T]

Y(z) z-m  y[-iT] zim

for m>0, m: const.

i 1

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The sample values y[iT] are the initial conditions already encountered in section 4.1 above.
This allows investigating the system response on an input signal in situations where the output signal from a previous stimulus has not
yet disappeared.

5.5 The Transfer Function of a System

The ratio Y(z)/X(z) is called the transfer function H(z) of a system. It is a unique specification of a system.
Hence Y(z)=H(z).X(z); the z-transform Y(z) of the output signal y[nT] of such system can be calculated from the transfer function H(z)
and the Laplace transform X(z) of the input signal x[nT] of the system. As both, X(z) and H(z), are most likely rational functions, Y(z)
will become a rational function as well, and partial fraction expansion is a systematic approach to determine y[nT] for a given x[nT] and
a given system.
Besides being useful for calculating system responses, the transfer function provides information about the stability of a system and
allows to determine the frequency characteristics of a system (see next sections).
Several options exist to determine the transfer function of a causal LTI system:
The transfer function H(z) can be determined from the system equation according to the previous section:
b0  b1z-1  b2z-2  ...  bl-1z-(l-1)  blz-l
H(z)
a0  a1z-1  a2z- 2  ...  ak -1z-(k -1)  ak z-k
As above, for real systems all coefficients ai and bi are real values.

The transfer function H(z) can also be determined from the unit pulse response h[nT] of the system simply by H(z)=Z
Z{h[nT]}.
Furthermore the transfer function can be determined graphically from any block diagram given for the system. For this purpose the
complete block diagram is sketched in the z-domain (i.e. signals are denoted X(z), Y(z), instead of x[nT], y[nT], ) and each block
is given a content that corresponds to its transfer function (i.e. a delay block will be given the content of z-1 instead of D, as z-1 is the
transfer function of a delay block; constant values in a block remain unchanged).
Having done this, the following block operations may be performed to subsequently reduce the number of blocks (and thereby the
block diagram's complexity) until only one block is left. The content of that block is the transfer function H(z) of the system.
Block diagram operations (in the z-domain!):
two blocks (with content H1(z) and H2(z)) in series can be replaced by one block with content H1(z).H2(z).
two blocks (with content H1(z) and H2(z)) in parallel can be replaced by one block with content H1(z)+H2(z).
two blocks (with content H1(z) in forward and H2(s) in feedback direction) in a basic feedback arrangement can be replaced by one
H1(z)
. The sign is contrary to that of the feedback signal at the combination (sum point).
block with content
1 # H1(z) H2 (z)
A branch may be moved from before a block to behind that block and vice versa; a compensating block has to be added to the
block diagram so that the overall functionality remains unchanged.
A sum point may be moved from before a block to behind that block and vice versa; a compensating block has to be added to the
block diagram so that the overall functionality remains unchanged.
One should avoid swapping a branch and an adjacent sum point; the compensations needed to keep the overall functionality
unchanged usually introduce additional complexity into the block diagram instead of simplifying it.

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5.6 Stability and Frequency Response of a System

5.6.1 System Stability
The definition of the BIBO-stability as given before is not useful to determine whether or not a given system is stable. For this purpose
a set of stability criteria have been developed. Only one of them shall be described here. For stability analysis the initial conditions are
assumed to be 0.
Stability analysis based on the poles of the system
This provides complete information on a system's stability but requires that H(z) is given and all of its poles are exactly
known/determined.
Given a causal LTI system with a rational transfer function H(z)=N(z)/D(z), where N(z) is the numerator polynomial and D(z) is the
denominator polynomial.
The following procedure determines the stability characteristics of the system:
1. Determine all zeros Ei and polesDi of H(z) and specify H(z) in its factor form.
2. If N(z) and D(z) have a common factor (z-z0), both shall be divided by that factor, and only the resulting H(z) is investigated further.
3. The system is
stable if all poles Di of H(z) are inside the unit circle (i.e. |Di|<1).
marginally stable if H(z) has no poles outside the unit circle and only single-order poles on the unit circle.
instable in all other cases (multiple-order poles on unit circle and/or poles outside the unit circle).

5.6.2 System Frequency Response

The frequency response Hd(f) of a system specifies the behavior of a system with regard to input signals of different frequencies.
Hd(f) is a continuous function of the frequency f.
It sometimes is also expressed as function of the angular frequency Z; note however that the notation Hd(jZ) is misleading as it is not
produced from Hd(f) through replacing f by jZ but through replacing f by Z/2S.
For DT-systems Hd(f) is periodic with period fs; thus its trace is infinitely repeated over the frequency axis; Hd(f)=Hd(f-fs).

The frequency response is a complex valued function of f; it thus has an amplitude and a phase value for each f: Hd (f) | Hd (f) | e jHd (f) .
Its primary purpose is to find the output signal the system would produce when the input signal is a sinusoidal signal of frequency f0:
x[nT] = A.cos(2Sf0nT + 4).
An LTI system will respond with a sinusoidal output signal of same frequency:
y[nT] = B.cos(2Sf0nT + );
only amplitude and phase of y[nT] may change, compared to x[nT].
The frequency response specifies the amount of change occurring to amplitude and phase:
B=A.|Hd(f0)|
4+Hd(f0)
When graphically presenting the frequency response, usually two diagrams are required:
the amplitude response: a graph showing |Hd(f)| over f, and
the phase response: a graph showing Hd(f) over f.
As a DT system may only experience input signals with frequencies f<fs/2 due to Shannon's sampling theorem, the knowledge of one
period of Hd(f) is sufficient, and graphical representations often are reduced to such frequency interval.
Several options exist to determine the frequency response of a system
The so-called discrete time Fourier transform (DTFT) calculates Hd(f) from the unit pulse response h[nT] according to
f

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Hd (f)

h[nT] e- j 2Sn f T .

n -f

The DTFT is not further covered in this course.

A straight-forward method to determine Hd(f) in case H(z) is known is:
Hd (f) H(z)

z e j2 SfT

It assumes that Hd(f) exists, which is not further investigated here (Hd(f) exists for all systems that are BIBO-stable).
Note that Hd(f) obviously corresponds to H(z) when evaluating H(z) on the unit circle (the amplitude of z=ej2SfT is 1, and its phase is
2SfT; thus the phase grows with growing f, periodically repeating with 1/T=fs).
Note further that the system behavior for constant input signals (f=0) can be determined from H(z=1); for the highest possible
frequencies (ffs/2) it can be determined from H(z-1)

SIG_Compendium_01.doc / 24.03.14 / RBz

SIG Compendium

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