General

CTFS, Continuous Time Fourier Series, (Continuous Time, Periodic) CTFT, Continuous Time Fourier Transform, (Continuous Time, Nonperiodic)

(Synthesis Equation) (Analysis Equation) (Synthesis Equation) (Analysis Equation)

CTFT Properties
Linearity

Time and Freq

Shifting
Time and Freq
Scaling
Transform of a
Conjugate
Multiplication -
Convolution Duality

Time
Differentiation

Transforms of
Periodic Signals

Parseval’s Theorem The product of two energy signals in the time domain corresponds to the convolution of their transforms in the frequency domain.

Integral Definition
of an Impulse

Duality

Total-Area Integral
Using Fourier
Transforms
Integration

CTFT
The Fourier transform of a signal x(t) exists if (1) it is an energy signal and (2) any discontinuities are finite – This requires the trick of using limits
• True for all signals of finite amplitude and duration • As with CTFS, convergence does not imply that the inverse Fourier transform will recover the signal
• Do periodic signals have a Fourier transform? • However, will be equal at all points except for discontinuities
– No, but we can still apply the transform if we allow X(j!) to be expressed in terms of impulse functions
CTFS
Convergence Since the CTFS includes an infinite series, we must consider under what conditions it converges
• An infinite sum is said to converge so long as it is bounded A sufficient condition for convergence (not proven) is
• In other words, the signal has – Finite power – Finite energy over a single period

Dirichlet The Fourier series representation of a periodic signal x(t) converges if all of the following 3. Finite number of distinct maxima and minima in T
Conditions for conditions are met. 4. x(t) is single valued
Convergence These are sufficient, but not necessary, conditions.
1.
2. Finite number of discontinuities in a period T
Terminology The coefficients X[k] are called the spectral coefficients or Fourier series coefficients of x[n] The function argX[k] is called the phase spectrum of x(t)
The function |X[k]| is called the magnitude spectrum of x(t)
Complex • The phase is odd: X*−k+ = −X[k]
Conjugate • The real part is even: Re,X*−k+- = Re,X*k+-
Symmetry This complex-conjugate symmetry ensures that the imaginary components of the sum cancel • The imaginary part is odd: Im,X*−k+- = −Im,X*k+-
each other and x(t) is therefore real-valued.
The complex-conjugate symmetry of the coefficients ensures that for real-valued signals x(t)
• The amplitude is even: |X*−k+| = |X*k+|
Gibb's • For finite Fourier series estimates there is apparent overshoot • The maximum overshoot (error) does not decrease as N → ∞
Phenomenon • As N increases, the edges become sharper and more accurate
Signals
Instantaneous
Signal Power
Signal Energy

Average Signal
Power

Energy For many signals the energy integral does not converge because the signal is not "time limited" A signal cannot be both an energy signal and a power signal
Comments and therefore the energy goes to infinity.
A signal is called an energy signal if E∞ < ∞ Signals with finite energy have zero average power:
A signal is called a power signal if 0 < P ∞ < ∞ Signals of finite duration and amplitude have finite energy:
A signal can be an energy signal, a power signal, or neither type

Signals with finite average power have infinite energy:

Odd/Even A signal is even iff x(t) = x(−t) A signal is odd iff x(t) = −x(−t) cos(t) is an even signal sin(t) is an odd signal
Function Type Sum, Difference Product Quotient Derivative Integral
Both Even Even Even Even 1-Even Odd Odd + Constant
Both Odd Odd Even Even 1-Odd Even Even
1-Even, 1-Odd Neither Odd Odd
Eigenfunctions Any signal x(t) or x[n] that is only scaled when passed through a system is called an eigenfunction of the system The scaling constant c is called the system’s eigenvalue
and Complex exponentials are eigenfunctions of LTI systems
Eigenvalues , Complex sinusoids are the only eigenfunctions of LTI systems that have finite
power.
Continuous A signal x(t) is periodic if there exists a T > 0 such that x(t + T) = x(t) for all t
Time Periodic Fundamental period: the minimum value of T for which the above holds. Often denoted as T0.
Signals
Fundamental frequency: ,
Text Summary
Ch. 2 1. The term "continuous" and the term "continuous-time" mean different things. 3. A continuous-time impulse, although very useful in signal and system analysis is not a function in the
SOIP 2. Two signals that differ only at a finite number of points have exactly the same effect on any real physical ordinary sense.
system. 4. Many practical signals can be described by combinations of shifted and/or scaled standard functions
and the order in which scaling and shifting are done is significant.
Ch. 4 1. A system that is both homogeneous and additive is linear.
SOIP 2. A system that is both linear and time invariant is called an LTI system.
3. The total response of any LTI system is the sum of its zero-input and zero-state responses.
4. nonlinear systems can be analyzed with linear system tech. through an approx. called linearization.
Ch. 6 1. Every LTI system is completely characterized by its impulse response.
SOIP 2. The response of an LTI system to an arbitrary input signals can be found by convolving the input signal
with its impulse response.
3. Impulse response of a cascade conn of LTI systems is the convolution ofindividual impulse response.
Ch. 8 1. The Fourier series expresses a periodic signal as a sum of sinusoids at harmonics of the fundamental
SOIP frequency of the signal
2. The sinusoids used in the Fourier series to represent a signal are all orthogonal to each other.
3. The complex and trigonometric forms of the Fourier series are related through Euler's identity.
4. A Fourier series can be found for any signal that satisfies the Dirichlet conditions.
Ch. 10 1. The CTFS is a special case of the CTFT.; 2. A signal with an infinite period is aperiodic.
SOIP 3. Signals and systems are often more usefully described by their frequency-domain properties than their
time-domain properties.
4. The generalized CTFT, which allows impulses in the transform, includes periodic signals.
5. More a signal is localized in one domain (time or frequency), the less it is localized in the other domain.
General Tools
Complex
Numbers
5. A system is said to be BIBO stable if arbitrary bounded input signals always produce bounded output
signals.
6. All real physical systems are causal, although some may be conveniently and superficially described
as noncausal.
4. Impulse response of a parallel conn of LTI systems is the sum of the individual impulse response.
5. An LTI system is BIBO stable if its impulse response is absolutely integrable.
6. LTI systems can be represented by block diagrams, and this type of representation is useful both in
synthesizing systems and in understanding their dynamic behavior.
5. For continuous signals, the Fourier series converges exactly to the signal at every point.
6. For discontinuous signals, the Fourier series converges exactly to the signal at every point exept
points of discontinuity. The effects of the actual signal and its Fourier series representation on any real
physical system are the same.
7. If a stable LTI system is excited by a periodic signal, the response is also a periodic signal with the
same fundamental period.
6. Convolution and multiplication of functions are dual operations in the time and frequency domains.
7. The Fourier transform of a periodic signal consists only of impulses.
8. Signal energy is conserved in the Fourier transformation process.
9. Most Fourier transforms of signals with engineering usefulness can be done most efficiently using
the tables of transforms and the properties of the transform.

Chapter 3 - Mathematical Description of Discrete-Time Signals

A discrete-time signal can be formed from a continuous-time signal by sampling A discrete-time function is not defined at non integer values of discrete time
SOIP

Discrete-time signals formed by sampling periodic continuous-time signals may have a different period or may even be aperiodic
Two different-looking analytical descriptions of discrete-time functions may, in fact, be identical A time-shifted version of a discrete-time function can produce decimation or undefined
Time scaling a discrete-time function can produce decimation or undefined values, phenomena that do not occur when time scaling continuous-time functions.
Since discrete-time functions are only defined for integer values of n, the values of expressions line g[2.7] or g[3/4] are simply undefined.
If we create a discrete-time sinusoid by sampling a continuous-time sinusoid, the period of the discrete-time sinusoid may not be readily apparent and, the discrete-time sinusoid may not even be periodic.
Other Notes

Where,
And
The requirement on a discrete-time sinusoid that it be periodic is that, for some discrete time n and some integer m, 2F0n = 2m. In
words, F0 must be a rational number (ratio of integers).
Units Discussion: "n" is a really a time index, not time itself. F0 should have units of cycles/sample to make 2F0n dimensionless and 0 should have units of radians/sample.
Unit Impulse Disc Time
Functions

Kronecker Delta Unit-Sequence Impulse Sampling Signum

Signal

Property
Unit Ramp Rectangle Periodic Impulse
Ramp[n]= rectNw[n]=
Time Compression Time Scaling of the form nKn, where |K|>1 and K is an integer. Time compression for discrete-time functions is similar to time compression for continuous-time functions in that the
function seems to occur faster in time. But in the case of discrete time functions there is another effect called decimation.
Time Expansion If we want to graph g[n/2] for each integer value of n, we must assign a value to g[n/2] by finding the corresponding value in the original function definition. But when n is one, n/2 is
one-half and g[1/2] is not defined. We could simply leave those values undefined or we interpolate.
Properties

Differencing The operation on a discrete-tie signal that is analogous to the derivative. The first forward difference of the discrete-time function g[n] is g[n+1]-g[n]. The first backward difference of a
discrete time function is g[n]-g[n-1], which is the first forward difference of g[n-1].
Accumulation The discrete-time counterpart of integration is accumulation (or summation). The accumulation function is described by:
Periodic Functions The fundamental frequency is F0 = 1/N0 in cycles or 0=2/N0 in radians
Signal Energy Signal
Power

Chapter 5 - Discrete -Time System Properties

Discrete-time systems are usually modeled by difference equations. The solution methods for difference equations are very similar to the solution methods for differential equations.
SOI

One common use for difference equations is to approximate differential equations. The properties of discrete-time systems are basically the same as they are for continuous-time systems.
P

A discrete-time system is stable if all its eigenvalues are less than one in magnitude.
Just as for continuous-time systems, any time a discrete-time system can exhibit an unbounded response to a bounded excitation of any kind, it is classified as BIBO unstable system. So the stability of feedback
systems depends on the nature of the feedback.
Other
Notes

If we multiply the excitation of this system by any constant, the response is multiplied by the same constant, so this system is homogeneous. If we delay the excitation of this system by any time n0, we delay
the response by that same time. Therefore, this system is also time invariant. If we add any two signals to form the excitation of the system, the response is the sum of the responses that would have occurred
by applying the two signals separately. Therefore, this system is an LTI discrete-time system. This system also has a bounded response for any bounded excitation. it is also stable.
Discrete-time LTI systems are usually described by linear, constant-coefficient difference equations. The eigenfunctions of these equations are functions of the form z^n where z is a complex constant.
Chapter 7 - Time-Domain Analysis of Discrete-Time Systems
Every LTI system is completely characterized by its impulse response. The response of an LTI system to an arbitrary signal can be found by convolving the signal with the system impulse response.
SOIP

The impulse response of a cascade connection of LTI systems is the convolution of the individual impulse responses
The impulse response of a parallel connection of LTI systems is the sum of the individual impulse responses A LTI system is BIBO stable if its impulse response is absolutely summable.
LTI systems can be represented by block diagrams and this type of representation is useful both in synthesizing systems and in understanding their dynamic behavior.
Convolution: The excitation for any discrete-time system is made up of a sequence of impulses with different strengths, occurring at different times. Therefore, invoking linearity and time-invariance, the
response of a LTI system will be the sum of all the individual responses to the individual impulses.
Convolution The value of the response y at any discrete time n can be found by summing all the products of the excitation x at discrete times m
with the impulse response h at discrete time n-m for m ranging from negative to positive infinity. For an LTI system, the impulse
Sum response of the system is a complete description of how it responds to any signal.
Convolution

Compact Form Associativity

Distributivity Commutativity
If, Differencing Sum
If an overall function is the convolution of two component functions, then time shifting either, but not both, of the two component functions time shifts the overall function by the same amount.
For a convolution sum to converge, both signals being convolved must be bounded and at least one of them must be absolutely summable
If the time of the first element in x is nx0 and the time of the first element of h is nh0, the time of the first element of y is nx0+nh0. If the time of the last element in y ix n x1 + nh1. The length of x is nx1-nx0+1 and the
length of h is nh1-nh0+1. So the extent of y is in the range of nx0+nh0≤n<nx1+nh1 and it length is So the length of y is one less than the sum of
the lengths of x and h.
Stability The response of y[n] of a system x[n] is Then, if x[n] is bounded, we can say that y[n] is bounded, we can say that y[n] is bounded if h[n] is absolutely summable (and therefore also
bounded). That is, if is bounded. A system is BIBO stable if its impulse response is absolutely summable.
Chapter 9 - Discrete-Time Fourier Series
The Fourier series expresses a periodic signal as a sum of sinusoids at harmonics of the fundamental frequency of the signal. The convergence of the DTFS is exact at every point
O

P
S

The sinusoids used in the Fourier series to represent a signal are all orthogonal to each other. The complex trigonometric forms of the Fourier are related through Euler's identity.
 The DTFS of a signal is a finite summation because of the nature of discrete time. If a stable LTI system is excited by a periodic signal, the response is also a periodic signal with the same fund. period.
The fundamental concept of the discrete-time Fourier series is to find a way of expressing any arbitrary signal as a linear combination of complex sinusoids. Then we can take advantage of linearity and time-
invariance and find the response to each complex sinusoid one at a time and then add all those response.
The discrete-time Fourier series (DTFS) expresses arbitrary signals as linear combinations of complex sinusoids so we can use superposition to find the response of any LTI system to any arbitrary signal simply by
summing the responses to the individual complex sinusoids.
In discrete time, exact representation is always achieved with a finite number of sinusoids.
The k=Nf complex sinusoid and the k=0 complex sinusoid are identical functions (because n must be an integer) as are the k=Nf+1 and the k=1 complex sinusoids. This proves that when we add any integer
multiple (including negative integers) of the representation time Nf to the harmonic number of any particular discrete-time complex sinusoid, we get an identical discrete-time complex sinusoid. Therefore, any
range of consecutive harmonic numbers k exactly Nf in length is a complete set of complex sinusoids.
DTFS Formulas If we let n0=0 the DTFS
becomes the DFT:
DTFS Formulas

So the DFT and the DTFS are very similar, differing only by a scale constant N F if the choice of the first n in the DTFS summation is n0=0. ;
Summarizing, if a signal x[n] has a fundamental period N0, and NF is an integer multiple of N0, its DTFS harmonic function is:
where 0≤n<NF
In the very important special case in which we represent a periodic function by its DTFS
using its fundamental period N0 as the representation time, the forward and inverse
DTFS operations become:
If we use an integer multiple m of the fundamental period as the
representation time, the formulas become:
Time Time Linearit Frequency Conjugation For these
shifting reversal y shifting properties

Time Let z[n]=x[an], a>0. If a is not an integer, then some values of z[n] will be undefined and a
Scaling DTFS cannot be found for it. If as is an integer, the z[n] is a decimated version of x[n], and
Properties

some values of x[n] do not appear in z[n]. The result to the right says the at the harmonic
function of z is the same as the harmonic function for x except divided by m.
Change of Using two periods instead of one does not add any information because the signal is exactly
Period the same in each period. We would in general get extra harmonic information, but here all
of the extra harmonics have zero amplitude.

Multiplication and Convolution Duality

Let be the notation for a periodic convolution

First Backward Accumulation

Difference

For x[n] even and real-valued, X[k] is even and real-valued For x[n] odd and real-valued, X[k] is odd and purely imaginary
Parseval's theorem The average signal power of the signal is equal to the sum of the average signal powers in ;
its DTFS harmonics
Chapter 11 - Discrete Time Fourier Transform
The DTFS is a special case of the DTFT The DTFT is always periodic with period one in the F domain or period 2 in the domain
O

P
S

For periodic signals, there are simple conversions between a Fourier transform and a Fourier Series.
Defin
ed

DTFT computed at freq F=k/NF

Greenwood: "means DTFT is DFT if θ is restricted"; Greenwood: "DFT is a sampled DFT"
Linearity Time Frequenc
Shifting y Shifting
Time Scaling Transform of Time
conjugate Reversal
Differencing Accumulation

Multiplication- Accumulation
Convolution Definition of a
Duality periodic Impulse

Parseval's Theorem

The fundamental period is N0=1/F0=2/ "Handy"

Properties

"Normalized Analog World uses freq in Hz or cycles/sec

Frequency" Digital World uses normalized freq in Hz/fs; fs=sample frequency
DTFS gives a continuous function not in Hz bu in Hz/fs.
Normalized Frequency "Wraps Around"

Concept:
Remark #1 X(θ) is computed from samples of x(t) (i.e. from x*n+) but X(θ) is a continuous function
Remark #2 ;

Remark #3 X(θ) is periodic with period 2, 0 to 2, but in practice it's
calculated from - to +.
"What should fs be? " per "Nyquist Criteria" to prevent overlap distortion, let Fx be the highest frequency component of X(θ), then fs>Fx
DFT presumes x(t) is periodic, DTFT works on "aperiodic" signals, DFT gives sample values of DTFT only over certain time duration. This is some times called "windowing".
Mcnames Lower frequencies are those that are near 0, high frequencies are those near ±, intermediate frequencies are those in between. The highest frequency,  radians per sample is equal to 0.5 cycles per sample.
Miscellaneous

Series
Eqs.
Chapter 16 - The z transform
Some signals that do not have a DTFT do have a z transform Every z transform has an associated region of convergence in the z plane
SOI

An inverse z transform can be found by the direct inversion integral, synthetic division or partial-fraction expansion. Use of the direct inversion integral is rare and synthetic division does not provide a closed-
P

form result. Therefore, partial fraction expansion is usually preferred.

 It is possible to do analysis of discrete-time systems with the Laplace transform through the use of continuous-time impulses to simulate The unilateral z transform can be used to solve difference equations
discrete time. But the z transform is notationally convenient. with initial conditions
Discrete-Time systems can be described by difference equations or block diagrams in the time or frequency domain.
A discrete time LTI system is stable if all the finite poles of its transfer function lie in the open interior of the circle.
The three most important types of system interconnections are the cascade connection, the parallel connection and the feedback connection.
The unit sequence and sinusoid are important practical signals for testing system characteristics.
Except for a multiplicative constant, a system's frequency response can be determined directly from its pole-zero diagram.
Discrete-Time systems can closely approximate the actions of continuous-time systems and the approximation The Direct Form II, cascade and parallel realizations are important standard ways of
improves as the sampling rate is increased. realizing systems.
Greenwood Design a IIR Filter notes Greenwood Z-Transform Notes
We want to design a low-pass filter (LPF). Many tables exist that give the transfer Z-Transform to DTFT Relation: and if
function for “normalized” analog filters. For example, one such normalized analog
LPF is Normalized filters have a cutoff frequency of 1 Hz (i.e., Also, The DTFT -- Complex Sinusoids, Z-Transform -- Complex Exponentials.
We want our LPF to have a cutoff frequency of 150 Hz and a sample Note: 1- DTFT Assumes r=1; 2- The z transform "might" exist anywhere in the Z plane (must be in region of convergence);
frequency of 1.28Khz. We will use the impulse invariant method. First we must 3 - DTFT exists in the Z-Plane only on the unit circle.
frequency scale the normalized analog filter transfer function to get the correct Important: You can only substitute if and only if the ROC contains the unit circle in the Z-Plane
If the ROC of the Z-Transform contains the unit circle the DTFT exists.
cutoff frequency
Note: 1 - H(z) will only have poles at z=0 if H(z) has z^-1 terms (causal terms); 2- H(z) will only have poles at Z=∞ if there
are Z terms (non-causal); 3 - If then the ROC contains the unit circuit (DTFT
exists)
NOTE: The impulse invariant method only will work with an all-pole analog transfer Linear Phase: You must have linear phase to prevent distortion
The Laplace Transform is used for stability analysis for systems described by ODE's; The Z-Transform is the same thing for
function.
systems described by difference equations.
The two poles are at The
partial fraction expansion yields. Def(2-Sided Z Transform):
Now, we know: is the sample rate. This gives a Def(1 Sided Z Transform): , For causal x[n] (i.e. x[n]=0 For all n <0
Def(Region of Convergence) ROC: Values of Z where X(Z) converges
filter transfer function: Def(Pole): The Z-Value where H(z) = ∞; Def(Zero): The Z Value where H(z) = 0; In a Z-transform the number of poles =
number of zeros.
After substituting in the sample rate T, the A and B coefficients and the poles
Digital Filter: A discrete time system that will take samples of x(t) and filter.
Infinite Impulse Response (IIR), The transfer function has poles, there is no feedback
Then substituting to get Finite Impulse Response (FIR); have finite impulse response. The impulse response has a finite duration.
Notice at which is close to the sample frequency in Hz (1280).
Example 1: Example 2: Example 3: :
This large gain value is typical of IIR filters and can possibly cause overflow problems.
So it is common practice to multiply the filter gain by the sample rate (T).

What restriction on for DTFT to

exist? Answer:
The inverse Z transform becomes:

Exam 1
N samples of a signal x(t) are taken over one period to calculate a set of 10 Fourier coefficients using a DFT. How The spacing between Fourier Coefficients is is the sample frequency and N is the number
1
would the DFT results change if the N samples were zero padded with 3N additional samples? of samples. Zero padding with 3N samples makes the spacing
Four samples are taken over one period of a signal. The sample values are:
2 ; ; ;
x[0]=1; x[1]=0; x[2]=0; x[3]=0. Plot the first 5 Fourier Coefficients
3 A periodic signal g(t) is frequency limited. What effect does frequency limiting have on the CTFS? The CTFS of g(t) will be finite.
In the following LTI system ;
4
If
Given a) ; Thus,
5 a) Find all of the Fourier Series Coeffs for x(t).
b) The average power is
b) What is the average power in x(t)?
Briefly explain how the time compression of x(t) effects X(ω). The Fourier transform property is
6
Illustrate the effect by plotting the Fourier transform of sin(t) and the Fourier transform of sin(5t). Thus,

7 Let ; Find X(ω)

Consider the pulse train. The CTFS will produce a pulse train with near infinite slopes but you get overshoot and ringing at the corners. That overshoot and ringing
8 Briefly describe the Gibbs Phenomenon.
is the Gibbs phenomenon. Even taking an infinite CTFS will not eliminate it.
One of your classmates says "If the input to a LTI system Agree. The input is of the form: ;
can be represented with a CTFS, then the output can also But each term is an eigenfunction for a LTI system. Hence the same complex exponential appears on the system output. By superposition, a linear
9
be represented by a CTFS". Do you agree or disagree? combination of eigenfunctions on the input of a LTI system creates a linear combination of eigenfunctions on the output of the system. The output of the
Justify your answer. system is thus of the form: which is a CTFS.
For each of the following signals, indicate if they are power signals, energy signals or neither power or energy signals.
a) Therefore it is an energy signal.

10
b) ; Therefore it is neither

c) ; Therefore it is a power signal.

Exam 2
; Therefore, ; The 50-point DFT samples at 50 points around the unit circle.
1 Let Find the 50-point DFT for x[n].
That is at Thus, the DFT is given by
A speech signal is sampled at a 20,000 samples/second rate. A sequence of 1024 samples is used to compute a 1024-point DFT. a) What is the time a) ; b)
2
duration of this 1024 sample sequence? b) What is the frequency resolution (i.e., what is the frequency difference) between DFT samples?
For engineers the DFT is more useful than the Both the CTFS and the CTFT require solving an integral. This integral cannot be solved unless the equation for the signal is known. The DFT can be used to help create
3
CTFS or the CTFT. Why? the DTFS or the DTFT. The DFT only requires samples of the signal. Samples can always be taken without having to know anything about the signal's equation.
For each of the following find the fundamental period N0 a) The first signal is periodic with period 8 and the second signal with period 10. The least common multiple of 8 and 10 is N0=40.
4 a) b) The fundamental period is Then, where m and N0 are integers. . . The smallest
b) integer N0 is given when m=10. Thus, N0=9.
Explain the relationship between (a) the The DFT computes coeffs equal to the DTFS Fourier coefficients to within a scale factor. That is, the DFT computed coefficients are N times the DTFS Fourier coefficients.
5 DFT and DTFS and (b) the DFT and the
If then the DFT computes sampled values of X(θ). The samples are taken at Where N is the number of samples
DTFT
The DTFS of a periodic sequence x[n] has been computed. The Fourier coefficients are: ; This gives: x[0]=(X[0]+X[1]+X[2]+X[3])=0;
6
X[0]=3/2; X[1]=-1/2 +j1/2; X[2]=-1/2; X[3]=-1/2-j1/2, Find x[n] for n=0,1,2,3 x[1]=(X[0]+jX[1]-X[2]-jX[3])=1; x[2]=(X[0]-X[1]+X[2]-X[3])=2; x[3]=(X[0]-jX[1]-X[2]+jX[3])=3
7 The impulse response of a LTI is Find  Therefore, 