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In each of the above examples there is an input and an output, each of which is a time-varying
()
signal. We will treat a signal as a time-varying function, x t . For each time t, the signal has some
() ()
value x t , usually called x of t. Sometimes we will alternatively use x t to refer to the entire
signal x, thinking of t as a free variable.
() []
In practice, x t will usually be represented as a finite-length sequence of numbers, x n , in
which n can take integer values between 0 and N 1
, and where N is the length of the sequence.
[]
This discrete-time sequence is indexed by integers, so we take x n to mean the nth number in
sequence x, usually called x of n for short.
[] ()
The individual numbers in a sequence x n are called samples of the signal x t . The word
sample comes from the fact that the sequence is a discretely-sampled version of the continuous
signal. Imagine, for example, that you are measuring membrane potential (or just about anything
else, for that matter) as it varies over time. You will obtain a sequence of measurements sampled
at evenly spaced time intervals. Although the membrane potential varies continuously over time,
you will work just with the sequence of discrete-time measurements.
It is often mathematically convenient to work with continuous-time signals. But in practice,
you usually end up with discrete-time sequences because: (1) discrete-time samples are the only
things that can be measured and recorded when doing a real experiment; and (2) finite-length,
discrete-time sequences are the only things that can be stored and computed with computers.
In what follows, we will express most of the mathematics in the continuous-time domain. But
the examples will, by necessity, use discrete-time sequences.
Pulse and impulse signals. The unit impulse signal, written (t), is one at t = 0, and zero
everywhere else: (
(t) = 1 if t = 0
0 otherwise
The impulse signal will play a very important role in what follows.
One very useful way to think of the impulse signal is as a limiting case of the pulse signal,
(t): ( 1
(t) = if 0 < t <
0 otherwise
The impulse signal is equal to the pulse signal when the pulse gets infinitely short:
2
Unit step signal. The unit step signal, written u (t), is zero for all times less than zero, and 1
for all times greater than or equal to zero:
(
u(t) = 0 if t<0
1 if t0
P
Summation and integration. The Greek capital sigma, , is used as a shorthand notation
for adding up a set of numbers, typically having some variable take on a specified set of values.
Thus:
X5
i=1+2+3+4+5
i=1
P
The notation is particularly helpful in dealing with sums over discrete-time sequences:
3
X
x[n] = x[1] + x[2] + x[3]:
n=1
Arithmetic with signals. It is often useful to apply the ordinary operations of arithmetic to
=
signals. Thus we can write the product of signals x and y as z xy , meaning the signal made up
of the products of the corresponding elements:
3
= + + + +
... ...
Representing signals with impulses. Any signal can be expressed as a sum of scaled and
~( )
shifted unit impulses. We begin with the pulse or staircase approximation x t to a continuous
()
signal x t , as illustrated in Fig. 1. Conceptually, this is trivial: for each discrete sample of the
original signal, we make a pulse signal. Then we add up all these pulse signals to make up the
approximate signal. Each of these pulse signals can in turn be represented as a standard pulse
scaled by the appropriate value and shifted to the appropriate place. In mathematical notation:
1
X
x~(t) = x(k) (t k) :
k= 1
As we let approach zero, the approximation x ~(t) becomes better and better, and the in the limit
()
equals x t . Therefore,
1
X
x(t) = lim
!0
x(k) (t k) :
k= 1
Also, as ! 0, the summation approaches an integral, and the pulse approaches the unit impulse:
Z1
x(t) = x(s) (t s) ds: (1)
1
In other words, we can represent any signal as an infinite sum of shifted and scaled unit impulses. A
digital compact disc, for example, stores whole complex pieces of music as lots of simple numbers
representing very short impulses, and then the CD player adds all the impulses back together one
after another to recreate the complex musical waveform.
This no doubt seems like a lot of trouble to go to, just to get back the same signal that we
originally started with, but in fact, we will very shortly be able to use Eq. 1 to perform a marvelous
trick.
Linear Systems
A system or transform maps an input signal x (t) into an output signal y(t):
y (t) = T [x(t)];
where T denotes the transform, a function from input signals to output signals.
Systems come in a wide variety of types. One important class is known as linear systems. To
see whether a system is linear, we need to test whether it obeys certain rules that all linear systems
obey. The two basic tests of linearity are homogeneity and additivity.
4
Homogeneity. As we increase the strength of the input to a linear system, say we double it,
then we predict that the output function will also be doubled. For example, if the current injected
to a passive neural membrane is doubled, the resulting membrane potential fluctuations will double
as well. This is called the scalar rule or sometimes the homogeneity of linear systems.
Additivity. Suppose we we measure how the membrane potential fluctuates over time in
()
response to a complicated time-series of injected current x1 t . Next, we present a second (differ-
()
ent) complicated time-series x2 t . The second stimulus also generates fluctuations in the mem-
brane potential which we measure and write down. Then, we present the sum of the two currents
( )+ ( )
x1 t x2 t and see what happens. Since the system is linear, the measured membrane potential
fluctuations will be just the sum of the fluctuations to each of the two currents presented separately.
Superposition. Systems that satisfy both homogeneity and additivity are considered to be
linear systems. These two rules, taken together, are often referred to as the principle of superposi-
tion. Mathematically, the principle of superposition is expressed as:
Homogeneity is a special case in which one of the signals is absent. Additivity is a special case in
which = =1 .
Shift-invariance. Suppose that we inject a pulse of current and measure the membrane po-
tential fluctuations. Then we stimulate again with a similar pulse at a different point in time, and
again we measure the membrane potential fluctuations. If we havent damaged the membrane with
the first impulse then we should expect that the response to the second pulse will be the same as
the response to the first pulse. The only difference between them will be that the second pulse has
occurred later in time, that is, it is shifted in time. When the responses to the identical stimulus
presented shifted in time are the same, except for the corresponding shift in time, then we have
a special kind of linear system called a shift-invariant linear system. Just as not all systems are
linear, not all linear systems are shift-invariant.
In mathematical language, a system T is shift-invariant if and only if:
Convolution
Homogeneity, additivity, and shift invariance may, at first, sound a bit abstract but they are very
useful. To characterize a shift-invariant linear system, we need to measure only one thing: the way
the system responds to a unit impulse. This response is called the impulse response function of
the system. Once weve measured this function, we can (in principle) predict how the system will
respond to any other possible stimulus.
5
Impulses
Impulse Impulse Response
For
example
The way we use the impulse response function is illustrated in Fig. 2. We conceive of the input
stimulus, in this case a sinusoid, as if it were the sum of a set of impulses (Eq. 1). We know the
responses we would get if each impulse was presented separately (i.e., scaled and shifted copies of
the impulse response). We simply add together all of the (scaled and shifted) impulse responses to
predict how the system will respond to the complete stimulus.
Now we will repeat all this in mathematical notation. Our goal is to show that the response (e.g.,
membrane potential fluctuation) of a shift-invariant linear system (e.g., passive neural membrane)
can be written as a sum of scaled and shifted copies of the systems impulse response function.
The convolution integral. Begin by using Eq. 1 to replace the input signal x (t) by its repre-
sentation in terms of impulses:
Z 1
y (t) = T [x(t)] = T x(s) (t s) ds
2 1 3
1
X
= T lim
4
!0 k = 1
x(k) (t k) 5:
Using additivity,
1
X
y (t) = lim
!0
T [x(k) (t k) ]:
k= 1
Taking the limit, Z1
y (t) = T [x(s) (t s) ds]:
1
6
past present future
0 0 0 1 0 0 0 0 0 input (impulse)
0 0 0 1 1 1 1 1 1 input (step)
Using homogeneity, Z1
y (t) = x(s) T [ (t s)] ds:
1
()
Now let h t be the response of T to the unshifted unit impulse, i.e., h (t) = T [(t)]. Then by using
shift-invariance, Z 1
y (t) = x(s) h(t s) ds: (4)
1
Notice what this last equation means. For any shift-invariant linear system T , once we know its
()
impulse response h t (that is, its response to a unit impulse), we can forget about T entirely, and
()
just add up scaled and shifted copies of h t to calculate the response of T to any input whatsoever.
Thus any shift-invariant linear system is completely characterized by its impulse response h t . ()
The way of combining two signals specified by Eq. 4 is know as convolution. It is such a
widespread and useful formula that it has its own shorthand notation, . For any two signals x and
y , there will be another signal z obtained by convolving x with y ,
Z1
z (t) = x y = x(s) y (t s) ds:
1
Convolution as a series of weighted sums. While superposition and convolution may sound
a little abstract, there is an equivalent statement that will make it concrete: a system is a shift-
invariant, linear system if and only if the responses are a weighted sum of the inputs. Figure 3
shows an example: the output at each point in time is computed simply as a weighted sum of the
inputs at recently past times. The choice of weighting function determines the behavior of the
system. Not surprisingly, the weighting function is very closely related to the impulse response of
the system. In particular, the impulse response and the weighting function are time-reversed copies
of one another, as demonstrated in the top part of the figure.
7
Notes for Signals and Systems
Discrete-time systems that are linear and time invariant often are referred to as LTI systems. LTI
systems comprise a very important class of systems, and they can be described by a standard
mathematical formalism. To each LTI system there corresponds a signal h[n] such that the input-
output behavior of the system is described by
y[n] = x[k ]h[n k ]
k =
This expression is called the convolution sum representation for LTI systems. In addition, the
sifting property easily shows that h[n] is the response of the system to a unit-pulse input signal.
That is, for x[n] = [n],
y[n] = x[k ]h[n k ] = [k ]h[n k ] = h[n]
k = k =
Derivation It is straightforward to show that a system described by the convolution sum, with
specified h[n], is a linear and time-invariant system. Linearity is clear, and to show time
invariance, consider a shifted input signal x[n] = x[n no ] . The system response to this input
signal is given by
y[n] = x[k ] h[n k ]
k =
= x[k no ] h[n k ]
k =
To rewrite this expression, change the summation index from k to l = k N, to obtain
y[n] = x[l ] h[n no l ]
l =
= y[n no ]
This establishes time invariance.
It is less straightforward to show that essentially any LTI system can be represented by the
convolution sum. But the convolution representation for linear, time-invariant systems can be
developed by adopting a particular representation for the input signal and then enforcing the
properties of linearity and time invariance on the corresponding response. The details are as
follows.
Often we will represent a given signal as a linear combination of basis signals that have certain
desirable properties for the purpose at hand. To develop a representation for discrete-time LTI
systems, it is convenient to represent the input signal as a linear combination of shifted unit pulse
signals: [n], [n 1], [n + 1], [n 2], . Indeed it is easy to verify the expression
50
x[n] = x[k ] [n k ]
k =
Here the coefficient of the signal [n k ] in the linear combination is the value x[k]. Thus, for
example, if n = 3, then the right side is evaluated by the sifting property to verify
x[k ] [3 k ] = x[3]
k =
We can use this signal representation to derive an LTI system representation as follows. The
response of an LTI system to a unit pulse input, x[n] = [n], is given the special notation y[n] =
h[n]. Then by time invariance, the response to a kshifted unit pulse, x[n] = [n k ] is
y[n] = h[n k ] . Furthermore, by linearity, the response to a linear combination of shifted unit
pulses is the linear combination of the responses to the shifted unit pulses. That is, the response to
x[n], as written above, is
y[n] = x[k ] h[n k ]
k =
Thus we have arrived at the convolution sum representation for LTI systems. The convolution
representation follows directly from linearity and time invariance no other properties of the
system are assumed (though there are some convergence issues that we have ignored). An
alternate expression for the convolution sum is obtained by changing the summation variable
from k to l = n k:
y[n] = h[l ]x[n l ]
k =
It is clear from the convolution representation that if we know the unit-pulse response of an LTI
system, then we can compute the response to any other input signal by evaluating the convolution
sum. Indeed, we specifically label LTI systems with the unit-pulse response in drawing block
diagrams, as shown below
The demonstration below can help with visualizing and understanding the convolution
representation.
Response Computation
Evaluation of the convolution expression, given x[n] and h[n], is not as simple as might be
expected because it is actually a collection of summations, over the index k, that can take
different forms for different values of n. There are several strategies that can be used for
evaluation, and the main ones are reviewed below.
Analytic evaluation When x[n] and h[n] have simple, neat analytical expressions,
and the character of the summation doesnt change in complicated ways as n changes, sometimes
y[n] can be computed analytically.
51
Example Suppose the unit pulse response of an LTI system is a unit ramp,
h[n] = r[n] = n u[n]
To compute the response of this system to a unit-step input, write the convolution representation
as
y[n] = x[k ]h[n k ] = u[k ](n k ) u[n k ]
k = k =
= (n k ) u[n k ]
k =0
Note that in the second line the unit-step u[k] in the summand is removed, but the lower limit of
the sum is raised to zero, and this is valid regardless of the value of n. Now, if n < 0, then the
argument of the step function in the summand is negative for every k 0. Therefore y[n] = 0 for n
< 0. But, for n 0, we can remove the step u[n k ] from the summand if we lower the upper
limit to n. Then
n
y[n] = (n k ) = n + (n 1) + + 2 +1+ 0
k =0
Using the very old trick of pairing the n with the 0 , the (n 1) with the 1 , and so on, we see that
each pair sums to n. Counting the number of pairs for even n and for odd n gives
n(n + 1)
, n0
y[n] = 2
0, n<0
or, more compactly,
n(n + 1)
y[n] = u[n]
2
52
y[n] = x[k ]h[n k ]
k =
first plot h[k] as shown,
Then, for the value of n of interest, flip and shift. For n = 3, we plot h[3 k] and,
to facilitate the multiplication, plot x[k] immediately below:
y[3] = 3 + 2 + 1 = 6
To compute, say, y[4], slide the plot of h[3 k] one sample to the right to obtain a plot of h[4 k]
and repeat the multiplication with x[k] and addition. In simple cases such as this, there is little
need to redraw because the pattern is clear. Even in complicated cases, it is often easy to identify
ranges of n where y[n] = 0, because the plots of x[k] and h[n k] are non-overlapping. In the
example, this clearly holds for n < 0.
LTI cleverness The third method makes use of the properties of linearity and
time invariance, and is well suited for the case where x[n] has only a few nonzero values. Indeed,
it is simply a specialization of the approach we took to the derivation of the convolution sum.
Example With an arbitrary h[n], suppose that the input signal comprises three nonzero lollypops,
and can be written as
x[n] = [n] + 2 [n 1] 3 [n 3]
Then linearity and time invariance dictate that
y[n] = h[n] + 2h[n 1] 3h[n 3]
Depending on the form of the unit-pulse response, this can be evaluated analytically, or by
graphical addition of plots of the unit-pulse response and its shifts and amplitude scales.
Remark The convolution operation can explode fail to be well defined for particular choices
of input signal and unit-pulse response. For example, with x[n] = h[n] = 1, for all n, there is no
53
value of n for which y[n] is defined, because the convolution sum is infinite for every n. In our
derivation, we did not worry about convergence of the summation. This again is a consequence of
our decision not to be precise about classes of allowable signals, or, more mathematically,
domains and ranges. The diligent student must always be on the look out for such anomalies.
Furthermore, there are LTI systems that cannot be described by a convolution sum, though these
are more in the nature of mathematical oddities than engineering legitimacies. In any case, this is
the reason we say that essentially any LTI system can be described by the convolution sum.
The convolution of two signals yields a signal, and this obviously is a mathematical operation a
sort of weird multiplication of signals. This mathematical operation obeys certain algebraic
properties, and these properties can be interpreted as properties of systems and their
interconnections.
To simplify matters, we adopt a shorthand star notation for convolution and write
y[n] = ( x h)[n] = x[k ]h[n k ]
k =
Note that since, for any n, the value of y[n] in general depends on all values of the signals x[n]
and h[n], we use the more general operator notation style, In particular, we do not write
y[n] = x[n] h[n] because of the temptation to conclude that, for example, y[2] = x[2] h[2] .
Using this result, there are two different ways to describe in words the role of the unit-pulse
response values in the input-output behavior of an LTI system. The value of h[n k] determines
how the nth value of the output signal depends on the k th value of the input signal. Or, the value
of h[q] determines how the value of y[n] depends on the value of x[n q].
54
The proof of this property is a messy exercise in manipulating summations, and it is omitted.
Of course, distributivity is a restatement of part of the linearity property of LTI systems and so no
proof is needed. The remaining part of the linearity condition is written in the new notation as
follows. For any constant b,
((bx) h)[n] = b ( x h)[n]
Shift Property This is simply a restatement of the time-invariance property, though the
notation makes it a bit awkward. For any integer no , if x[n] = x[n no ] , then
( x h ) [n] = ( x h ) [n no ]
Identity It is worth noting that the star operation has the unit pulse as an
identity element. Namely,
( x )[n] = x[n]
This can be interpreted in system-theoretic terms as the fact that the identity system, y[n] = x[n]
has the unit-pulse response h[n] = [n] . Also we can write ( ) [ n] = [ n] , an expression
that says nothing more than: The unit pulse is the unit-pulse response of the system whose unit-
pulse response is a unit pulse.
These algebraic properties of the mathematical operation of convolution lead directly to methods
for describing the input-output behavior of interconnections of LTI systems. Of course we use
block diagram representations to describe interconnections, but for LTI systems we label each
block with the corresponding unit-pulse response. For example,
55
Commutativity and associativity imply that the interconnections
as follows, in an attempt to obtain a description for its input-output behavior. With the
intermediate signal e[n] labeled as shown, we can write
y[n] = ( g e)[n]
and
e[n] = x[n] (h y )[n]
as descriptions of the interconnection. Substituting the second into the first gives
y[n] = ( g x)[n] ( g h y )[n]
or, writing y[n] = ( y )[n] we get
( ( g h) y ) [n] = ( g x)[n]
However, we cannot solve for y[n] on the left side unless we know that the LTI system with
unit-pulse response ( g h)[n] is invertible. Lets stop here, and return to the problem of
describing the feedback connection after developing more tools.
But for systems without feedback, the algebraic rules for the convolution operation provide an
easy formalism for simplifying block diagrams. Typically it is easiest to start at the output signal
and write descriptions of the intermediate signals (labeled if needed) while working back toward
the input signal.
56
Example For the interconnected system shown below, there is no need to label internal signals as
the structure is reasonably transparent.
Since the input-output behavior of a discrete-time LTI system is completely characterized by its
unit-pulse response, h[n], via the convolution expression
y[n] = x[k ]h[n k ] = h[k ]x[n k ]
k = k =
the input-output properties of the system can be characterized very precisely in terms of
properties of h[n].
Causal System An LTI system is causal if and only if h[n] = 0 for n < 0, that is,
if and only if h[n] is right sided.
The proof of this is quite easy from the convolution expression. If the unit-pulse response is right
sided, then the convolution expression simplifies to
y[n] = h[k ]x[n k ]
k =0
and, at any value of n, the value of y[n] depends only on the current and earlier values of the input
signal. If the unit-pulse response is not right sided, then it is easy to see that the value of y[n] at a
particular n depends on future values of the input signal.
57
Memoryless System An LTI system is memoryless if and only if h[n] = c[n], for
some constant c. Again, a proof is quite easy to argue from the convolution expression.
To prove this, suppose x[n] is a bounded input, that is, there is a constant M such that | x[n] | M
for all n. Then the absolute value of the output signal satisfies
| y[n] | = | h[k ]x[n k ] | | h[k ] || x[n k ] |
k = k =
M | h[k ] |
k =
Therefore, if the absolute summability condition holds, the output signal is bounded for any
bounded input signal, and we have shown that the system is stable.
To prove that stability of the system implies absolute summability requires considerable
cleverness. Consider the input x[n] defined by
1, h[n] 0
x[n] =
1, h[n] < 0
Clearly x[n] is a bounded input signal, and the corresponding response y[n] at n = 0 is
y[0] = h[k ] x[k ] = | h[k ] |
k = k =
Since the system is stable, y[n] is bounded, and therefore y[0] is bounded, and therefore the unit-
pulse response is absolutely summable.
58
Example To compute an inverse of the running summer, that is, the LTI system with unit pulse
response h[n] = u[n] , we must find hI [n] that satisfies
u[k ] hI [n k ] = [n]
k =
Simplifying the summation gives
hI [n k ] = [n]
k =0
It is clear that we should take hI [n] = 0 , for n < 0 . Using this result, for n = 0 the requirement
is
hI [k ] = hI [0] = 1
k =0
For n = 1 the requirement is
hI [1 k ] = hI [1] + hI [0] = 0
k =0
which gives hI [1] = 1 . Continuing for further values of n, it is clear that the inverse-system
requirement is satisfied by taking all remaining values of hI [n] to be zero. Thus the inverse
system has the unit pulse response
hI [n] = [n] [n 1]
Of course, it is easy to see that in general the output of this inverse system is the first difference of
the input signal.
The response of a DT LTI system to the basic singularity signals is quite easy to compute. If the
system is described by
y[n] = x[k ]h[n k ]
k =
then the unit pulse response is simply y[n] = h[n] . If the input signal is a unit step, x[n] = u[n] ,
then
y[ n] = u[k ]h[n k ] = h[n k ]
k = k =0
n
= h[l ]
l =
In words, the unit-step response is the running sum of the unit-pulse response. Of course, if the
system is causal, that is, the unit-pulse response is right sided, then
59
Impulse Response
The impulse response of a linear system h (t) is the output of the system
at time t to an impulse at time . This can be written as
h = H( )
(t) h(t, 0)
0 t 0 t
H
(t ) h(t, )
0 t 0 t
t on the left is a specific value for time, the time at which the output
is being sampled.
t on the right is varying over all real numbers, it is not the same t as
on the left.
The output at time specific time t on the left in general depends on
the input at all times t on the right (the entire input waveform).
h = h0 = H(0 ).
First, (2) is something like zero, so H(0) would be zero. Second, the
value of h(2) depends on the entire input waveform, not just the
value at t = 2.
H 0
(t)
(2) h(t, 0) h(2, 0)
0 2 t 0 2 t
Time-invariance
If H is time invariant, delaying the input and output both by a time
should produce the same response
h (t) = h(t ).
(t) h(t)
0 t 0 t
H
(t )
h(t )
0 t 0 t
ay = S(ax).
y1 + y2 = S(x1 + x2 ).
Extended Linearity
x y
H
y (t) = H (x(t))
Z
= H x( ) (t)d
If the system obeys extended linearity we can interchange the order of the
system operator and the integration
Z
y (t) = x( )H ( (t)) d.
h (t) = H( (t)).
Input Output
(t) h(t)
0 t 0 t
(t ) h(t )
0 t 0 t
(x()d)(t ) (x()d)h(t )
x(t)
0 t 0 t
y(t)
x(t)
0 t 0 t
Z Z
x(t) = x()(t )d y(t) = x()h(t )d
x +
+
Z y
-
Solving the system equation tells us the output for a given input.
The output consists of two components:
The zero-input response, which is what the system does with no input
at all. This is due to initial conditions, such as energy stored in
capacitors and inductors.
x(t) = 0 y(t)
0 t 0 t
H
x(t) y(t)
0 t 0 t
H
x(t) = 0 y(t) = 0
0 t 0 t
H
Example: Solve for the voltage across the capacitor y (t) for an arbitrary
input voltage x(t), given an initial value y (0) = Y0 .
i(t) R
+
x(t) +
C y(t)
Simplifying
1 t/RC
A0 (t) = x(t) e
RC
which can be integrated from t = 0 to get
Z t
1 /RC
A(t) = x( ) e d + A(0)
0 RC
RC Circuit example
The impulse response of the RC circuit example is
1 t/RC
h(t) = e
RC
The response of this system to an input x(t) is then
Z t
y (t) = x( )h (t)d
Z0 t
1 (t )/RC
= x( ) e d
0 RC
ergy resolution and light collection efficiency were measured with single
htguide elements. Photon
From: Doshi et al, Med Phys. 27(7), p1535 July 2000
FIG. 5. A picture of the assembled detector module consisting of a 9!9
array of 3!3!20 mm3 LSO crystals coupled through a tapered optical fiber
ding the PMT socket containing the dynode resistor chain bundle to a Hamamatsu R5900-C8 PS-PMT.
s network, is 3 cm long, 3 cm wide, and 9.75 cm long.
These are used in Positiron Emmision Tomography (PET) systems.
were defined. The detectors were then configured in coinci-
METHODSDETECTOR CHARACTERIZATION
dence, 15 cm apart, and list-mode data was acquired by step-
Input is a sequence of impulses
Flood source histogram ping (photons).
a 1 mm diameter 22Na point source !same as used in
A detector module was uniformly irradiated with a 68Ge Sec. III C# between the detectors in 0.254 mm steps. The
nt source !2.6 " Ci#. The signals from the PS-PMT were point source was scanned across the fifth row of the detector.
ated and digitized as described above in Sec. II D. The For each opposing crystal pair, the counts were recorded as a
wer energy threshold was set (Lecture
Cuff to approximately
3) $100 keV function
ELE 301: of the point
Signals source position. A lower energyFall
and Systems win-
2011-12 21 / 55
h the aid of the threshold on the constant fraction dis- dow of $100 keV was applied. The FWHM of the resulting
minator and no upper energy threshold was applied. distribution for each crystal pair was determined to give the
intrinsic spatial resolution of the detectors.
Energy spectra Output is superposition of impulse responses (light).
E. Detector efficiency
Boundaries were drawn on the 2D position map to define
ook-up table !LUT# which relates position in the 2D his- A measure of the absolute detector efficiency was ob-
ram to the appropriate element in the LSO array. The raw tained. A 18F point source with known activity !68 " Ci# was
Input: Photons
mode data were then resorted and a histogram of total Output: Light
placed 15 cm away from the face of the detector module. The
se amplitudes !sum of the four position outputs# gener- actual gamma-ray flux impinging on the detector face was
d for each crystal in the array. These energy spectra were calculated from the solid angle subtended by the detector
alyzed to determine the FWHM and the location of the module at the source. The constant fraction discriminator
1 keV photopeak of each crystal. These two parameters was set to eliminate electronic noise ($100 keV# and the full
asure the energy resolution and light collection efficiency, energy spectrum was obtained for each crystal over a fixed
pectively. time. A background measurement without the 18F point
t
source was also obtained to subtract the LSO background t
Timing resolution from the measurement. A lower energy window of 350 keV
was applied to all of the crystals and the number of counts
Two detectors were aligned facing each other, 15 cm
falling under the photopeak was calculated. The number of
art, and connected in coincidence. A 22Na point source
counts detected was then divided by the total number of
8 " Ci# encapsulated in a 25.4 mm diameter, 3 mm thick
gamma rays impinging on the detector module to obtain the
ar plastic disc with the activity in the central 1 mm was t
detector efficiency. t
ced in the center of the two detectors. For each detected
ncidence event, the sum of the four position signals for
h detector was sent to a constant fraction discriminator IV. RESULTSDETECTOR CHARACTERIZATION
ich generated timing pulses. The lower energy threshold A. Flood source histogram results
the CFD was set to approximately 100 keV. These two
An image of the flood histogram from one detector mod-
ing pulses !one for each module# were in turn fed into a
ibrated time-to-amplitude converter !TAC# module. The
t
ule is shown in Fig. 6. All 81 crystals from the 9!9 LSO t
array are clearly visible. An average peak-to-valley ratio of
put from the TAC was then digitized to produce the tim-
3.5 was obtained over the central row of nine crystals. Not
spectrum.
all crystals are uniformly spaced in the flood histogram after
applying Anger logic. This may be a result of the nonuni-
Coincidence point spread function
form tapering of the optical fiber taper, the nonuniform pack-
Flood source histograms of both detectors were obtained ing of the reflectance powder between the crystals, or most
Cuff (Lecture 3) ELE 301: Signals and Systems Fall 2011-12 22 / 55
Summary
Since h (t) = 0 for t < , we can replace the upper limit of the integral
by t Z t
y (t) = x( )h (t) d.
x(t) = A cos(2f1 t + )
produces an output
Z
y (t) = h( ) [A cos(2f1 (t ) + )] d.
where
Z
Hc (f1 ) = h( ) cos(2f1 )d
Z
Hs (f1 ) = h( ) sin(2f1 )d
Convolution Integral
The convolution of an input signal x(t) with and impulse response h(t) is
Z
y (t) = x( )h(t ) d
= (x h)(t)
or
y = x h.
This is also often written as
<t >t
Does not
x(t) y(t) contribute to y(t)
0 t 0 t
Z
! t
x(t) = x()(t )d y(t) = x( )h(t )d
If x(t) is also causal, x(t) = 0 for t < 0, and the integral further simplifies
Z t
y (t) = x( )h(t ) d.
0
Does not
contribute to y(t) <t >t
Does not
x(t) y(t) contribute to y(t)
0 t 0 t
! t ! t
x(t) = x( )(t )d y(t) = x( )h(t )d
0 0
t t
t t
Then integrate over to find y (t) for this t.
2
x()
1
-1 0 1 2 3
1 h()
-1 0 1 2 3
2
h()
1
-1 0 1 2 3
2
h(t )
1
-1 0 1 2 3
2
x() x()
h(t ) t <0 h(t )
1 1 0<t <1
-1 0 1 2 3 -1 0 1 2 3
2
x() x()
h(t ) 1<t <2 2<t <3
1 1 h(t )
-1 0 1 2 3 -1 0 1 2 3
y(t) = (x h)(t)
2 2
x()
1 h(t ) t >3 1
-1 0 1 2 3 -1 0 1 2 3
x(t) y(t)
h(t)
Impulse response:
1.5
1 h(t)
h
0.5
0
0 2 4 6 8 10
tt
1
x(t)
0.5
u
0
0 2 4 6 8 10
tt
1
y(t)
0.5
y
0
0 2 4 6 8 10
tt
x(t)
1
0.5
u
0
0 2 4 6 8 10
t
1 y(t)
0.5
y
0
0 2 4 6 8 10
tt
0 1 2 0 1 2 0 1 2
1 1 1
0 1 2 0 1 2 0 1 2
(t 1)
1 1 1
0 1 2 0 1 2 0 1 2
1 1 1
0 1 2 0 1 2 0 1 2
-1 0 1 2 3 -1 0 1 2 3
xh 2 hx 2
x()
h(t ) x(t )
1 1
h()
-1 0 1 2 3 -1 0 1 2 3
y(t) = (x h)(t)
2
-1 0 1 2 3
f g h = f h g = = h g f
Linearity
Convolution is also distributive,
f (g + h) = f g + f h
Convolution systems are linear: for all signals x1 , x2 and all , <,
h (x1 + x2 ) = (h x1 ) + (h x2 )
x1 (t) = x(t T )
y1 (t) = y (t T ).
x w y
f g
x y
( f g)
x w y
f g
x v y
g f
0 t 0 t
h
s(t)
u(t)
0 t 0 t
h
s(t)
u(t) h(t)
0 t 0 t 0 t
d
h
dt
(t)
u(t) h(t)
0 t 0 t 0 t
d
h
dt
Convolution
Convolution is a mathematical operation used to express the relation between input and output of
an LTI system. It relates input, output and impulse response of an LTI system as
x t = input of LTI
Continuous convolution
Discrete convolution
Continuous Convolution
Discrete Convolution
Deconvolution
Deconvolution is reverse process to convolution widely used in signal and image processing.
Properties of Convolution
Commutative Property
x1 (t) x2 (t) = x2 (t) x1 (t)
Distributive Property
x1 (t) [x2 (t) + x3 (t)] = [x1 (t) x2 (t)] + [x1 (t) x3 (t)]
Associative Property
Shifting Property
Scaling Property
Differentiation of Output
dy(t) dh(t)
dt
= x(t) dt
Note:
Here, we have two rectangles of unequal length to convolute, which results a trapezium.
1 + 2 < t < 2 + 2
3 < t < 4
Hence the result is trapezium with period 7.
Ay = Ax Ah
DC Component
DC component of any signal is given by
area of the signal
DC component = period of the signal
Ex: what is the dc component of the resultant convoluted signal given below?
= 3 4 = 12
Duration of the convoluted signal = sum of lower limits < t < sum of upper limits
= -3 < t < 4
Period=7
area of the signal
Dc component of the convoluted signal = period of the signal
Dc component = 12
7
Discrete Convolution
Let us see how to calculate discrete convolution:
Note: if any two sequences have m, n number of samples respectively, then the resulting
convoluted sequence will have [m+n-1] samples.
= [-1, 0, 3, 10, 6]
Here x[n] contains 3 samples and h[n] is also having 3 samples so the resulting sequence having
3+3-1 = 5 samples.
Periodic convolution is valid for discrete Fourier transform. To calculate periodic convolution all
the samples must be real. Periodic or circular convolution is also called as fast convolution.
If two sequences of length m, n respectively are convoluted using circular convolution then
resulting sequence having max [m,n] samples.
Ex: convolute two sequences x[n] = {1,2,3} & h[n] = {-1,2,2} using circular convolution
= [-1, 0, 3, 10, 6]
Here x[n] contains 3 samples and h[n] also has 3 samples. Hence the resulting sequence obtained
by circular convolution must have max[3,3]= 3 samples.
Now to get periodic convolution result, 1st 3 samples [as the period is 3] of normal convolution is
same next two samples are added to 1st samples as shown below:
Auto correlation
Cros correlation
Consider a signals xt . The auto correlation function of xt with its time delayed version is given by
R11 () = R() = x(t)x(t )dt [+ve shift]
= x(t)x(t + )dt [-ve shift]
F. T [R()] = ()
() = R()ej d
R() = x() x()
R(0) =
Auto correlation function of power signal 1 ,
|R()| R(0)
Auto correlation function and power spectral densities are Fourier transform pairs. i.e.,
F. T [R()] = s()
s() = R()ej d
R() = x() x()
Density Spectrum
Let us see density spectrums:
Consider two signals x1 t and x2 t . The cross correlation of these two signals R 12 () is given by
R12 () = x1 (t)x2 (t ) dt [+ve shift]
= x1 (t + )x2 (t) dt [-ve shift]
R12 () R21 ()
If R12 0 = 0 means, if x1 (t)x2 (t)dt = 0 , then the two signals are said to be orthogonal.
T
For power signal if limT 1 2
T x(t)x (t) dt then two signals are said to be orthogonal.
T
2
Cross correlation function corresponds to the multiplication of spectrums of one signal to the
complex conjugate of spectrum of another signal. i.e.
R12 () X1 ()X2 ()
This also called as correlation theorem.
Parsvel's Theorem
Parsvel's theorem for energy signals states that the total energy in a signal can be obtained by the
spectrum of the signal as
E= 1
2
|X()|2 d
Note: If a signal has energy E then time scaled version of that signal xat has energy E/a.