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Introduction (1A - Proakis)

1.1 Signals, Systems, and Signal Processing


*Signals - any physical quantity that varies with time, space or any other independent variable or variables
Examples:
Examples of Signals

Independent Variable

Speech Signal, Electrocardiogram (ECG),


Electroencephalogram (EEG)

Time

Image Signal

Spatial Coordinates

*Speech signal, ECG and EEG are normally expressed as a sum of sinusoids of different amplitudes,
frequencies and phases.
*System - responds to a stimulus or force associated with signal generation
*may also be defined as a physical device or software that performs an operation on a signal
*Signal source - stimulus in combination with the system
*Signal processing - linear or nonlinear operation on the signal
*Algorithm - method or set of rules for implementing the system by a program that performs the
corresponding mathematical operations.

1.1.1 Basic Elements of a Digital Signal Processing System


*Analog signal processing = both input and output are analog and analog are being processed as analog.

*Digital Signal processing - alternate method for processing the analog signal.
*Analog-to-Digital (A/D) converter - interface between the analog signal and the digital processor.
*Digital-to-Analog (A/D) converter - interface needed if the signal has to be provided in analog form.

1.1.2 Advantages of Digital over Analog Signal Processing


Advantages:
Advantages

Digital

Analog

1. Flexibility

Reconfiguring means simply changing


the program.

Reconfiguring means redesigning the


hardware altogether

2. Accuracy

Much better control of accuracy


requirements.

Due to tolerance of analog circuits,


control of accuracy is much more
difficult.

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difficult.
3. Storage

Can be easily stored on a magnetic


media (tape or disk) without loss of
signal fidelity.

Not possible.

4. Signal processing
through Algorithm
implementation

Can be done through a software on a


digital computer.

Very difficult to do.

5. Cheaper

Digital hardware is cheaper and


flexibility for modifications.

More expensive hardware.

Limitations of Digital: (1) speed of operation of A/D converters and digital signal processors. (2) signals
with extremely wide bandwidths
1.2 Classification of Signals
1.2.1 Multichannel and Multidimensional Signals
a. Scalar signals:

b. Vector signals - (also called as multichannel signals) generated by multiple sensors or sources
(e.g. ground acceleration due to earthquake)

One-dimensional: if the signal is a function of one variable


Multi-dimensional: if the signal is a function of multiple variables
1.2.2 Continuous Time and Discrete Time Signals
Classifications of signals depending on the characteristics of time variable and the values they take
1. Continuous Time Signals or analog signals - are defined for every value of time and they take on aaz
in the continuous interval (a, b) where a can be negative infinity and b can be positive infinity
2. Discrete Time Signals - defined only at certain specific values of time.

X (t)

Continuous time signal

X (n)

Discrete time signal

X (nT)

Discrete time signal with equally paced interval

In applications, discrete time signals may arise in two ways:


1. By sampling -selecting values of an analog signal at discrete-time instants.
2. By accumulating a variable over a period of time. (e.g. counting the number of cars using a given
street every hour, or recording the value of gold every day results in discrete time signals. )
1.2.3 Continuous Valued Versus Discrete-Valued Signals

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3. Continuous Valued signal - if a signal takes on all possible values on a finite or an infinite range
4. Discrete Valued signal - if a signal takes on values from a finite set of possible values
a. Usually equidistance and can be expressed in integer multiple
Digital signal - a discrete time signal having a set of discrete values
Quantization - the approximation process of converting a continuous valued signal into a discrete-valued
signal

1.2.4 Deterministic Versus Random Signals


Signal model - mathematical description of signals
Deterministic signals - any signal that can be uniquely described by an explicit mathematical expression , a
table of data or a well-defined rule. In other words, all past, present and future values of the signal are
known precisely and without uncertainty.
Random signals - cannot be described to a reasonable degree of accuracy by explicit mathematical
formulas or such a description is too complicated. Unpredictable (e.g. output of a noise generator, seismic
signal and speech signal)

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Introduction (1B - Proakis)


1.3 The Concept of Frequency in Continuous Time and Discrete-Time Signals
Frequency - closely related to a specific type of periodic motion called harmonic oscillation, which is
described by sinusoidal functions.
*inverse of time
1.3.1 Continuous-Time Sinusoidal Signals

Properties of Analog Sinusoidal Signals


1. For every fixed value of the frequency F, xa(t) is periodic.

2. Continuous-time sinusoidal signals with distinct (different frequencies) are themselves distinct.
3. Increasing the frequency F results in an increase
4. the rate of oscillation of the signal, in the sense the more periods are included in a given time
interval.
*Due to the continuity of the time variable t, we can increase the frequency F, without limit, with a
corresponding increase in the rate of oscillation.

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*Frequency is an inherently positive physical quantity (there no negative number of cycles per unit time),
but for mathematical convenience, negative frequencies are introduced.
*Phasors = two equal amplitude complex-conjugate exponential signals
Positive Frequency Counterclockwise uniform angular motion
Negative Frequency Clockwise angular motion
*Frequency range for analog sinusoids -->

1.32 Discrete-Sinusoidal Signals

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Characteristics of a Discrete-time sinusoid


1. It is periodic only if its frequency f is a rational number and if it follows:
where k and N are prime integers.
2. Discrete-time sinusoids whose frequencies are separated by an integer multiple of 2 are identical.

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Discrete Time Signals (Chapter 2.1 - Proakis)


2.1 Discrete-Time Signals
*Discrete time signal x(n) - function of an independent variable that is an integer n.
*It is not defined at instants between two successive samples.
*When n is a noninteger, x(n) is undefined, not zero.
*x(n) -> nth sample of the signal.
*If x(n) was obtained from sampling an analog signal, then xa(t) = xa(nT), where T is the sampling period (time
between successive samples).
Representations of Discrete-time signal or sequence:
1. Functional representation

2. Tabular representation
3. Sequence representation

4. Graphical representation

*Up arrow symbol indicated n=0 or origin.


*Ellipsis () indicates infinity.
2.1.1 Some Elementary Discrete-Time Signals
1. Unit Sample Sequence
(also called unit impulse)

2. Unit step signal


*Power Signal

3. Unit ramp signal


*Neither a power signal
nor an energy signal.

4. Exponential signal
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4. Exponential signal
*Power Signal

2.1.2 Classification of Discrete-Time Signals


Energy and Power Signals
Energy Signals
*E has to be finite.
*P = 0.
Power Signals
*Signals that possess infinite
energy but finite average power.

Periodic and Aperiodic Signals

Periodic

x(n+N) = x(n) for all n

Aperiodic

No value of N that can satisfy the equation.

*Fundamental period - smallest value of N


*Periodic signals are power signals
*If x(n) is a periodic signal with fundamental period N and takes on finite values, its power is given by:

Even or Odd Signals


Symmetric (Even) x(-n) = x(n)

Asymmetric (Odd) x(-n) = -x(n)

2.1.3 Simple Manipulations of Discrete-Time Signals

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Transformation of the independent variable (time)


Time Delaying

n = n-k, where k is a positive integer

Time Advancing

n = n-k, where k is a negative integer

Time Folding

n=-n (creating a reflection of the signal about the origin)

Time Scaling or Down-Sampling n=un, where u is an integer.


Amplitude Scaling

y (n) = Ax(n)

Sum of two signals

y (n) = x1 (n) + x2 (n)

Product of two signals

y (n) = x1 (n) * x2 (n)

*Not commutative.
Example:
x(n)

y (n) = x(2n)

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Discrete Time Systems


2.2 Discrete-Time Systems (Chapter 2.2 - Proakis)
*Discrete Time System - a device or an algorithm that performs some prescribed operation on a discrete-time signal to produce another
discrete time signal
Excitation - incoming discrete time signal
*Response - outcoming discrete time signal

*We say that the input x(n) is transformed by the system into a signal y(n), where T is the transformation or the operator

2.2.1 Input-Output Description of Systems


*Input-output description of a discrete-time system - consists of a mathematical expression or a rule, which explicitly defines the relations
between the input and output systems.
*Initial condition
*Initially relaxed system - no prior excitation
*Every system is relaxed at n is negative infinity, hence output is solely and uniquely determined by the given input.
2.2.2 Block Diagram Representation of Discrete-Time Systems
*memoryless operation - no need to store neither of the sequences in order to perform the operation
Adder

*memoryless operation

Constant Multiplier

*memoryless operation

Signal Multiplier

*memoryless operation

Unit Delay Element

*requires memory
*denoted by Z-1

Unit Advanced

*requires memory
*denoted by Z

Classification of Systems
1. Static versus dynamic systems
a. static (memoryless) - if its output at any instant n depends at most on the input sample at the same time, but not on past or
future samples of the input.
b. dynamic (memory)

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Static
Static
Dynamic
Dynamic

Dynamic

Sample Problems:
a.

Static

b.

Dynamic

c.

Static

d.

Dynamic

e.

Static

f.

Static

g.

Static

h.

Static

i.

Dynamic

j.

Dynamic

k.

Static

l.

Dynamic

m.

Static

Static

*Memory = Delay, Advanced, Folding, Internal Multiply and Summation


2. Time-invariant versus time-variant systems
a. Time-invariant - input-output characteristics do not change with time.
*Dapat pareho ang values kapag dinelay lang versus kinalculate.
b. Time variant - if there is a single value that disapproves it.
Time Invariant
s
Time Variant
*n also changes!
Time Variant
Time Variant

3. Linear and NonLinear


a. Linear system satisfies the superposition principle
*Superposition principle: requires that the response of the system to a weighted sum of signals be equal to the corresponding weight
sum of the responses (outputs) of the system to each of the individual input signals.
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sum of the responses (outputs) of the system to each of the individual input signals.
*Transformation of two signals should be equal to the individually transformed signals!
*Multiplicative or scaling property and additivity property
b. Non-linear - superposition test failed and zero input zero output failed!
Linear

Linear
Non-Linear
Linear
Non Linear (Failed the zero input-zero output test)
4. Causal versus Noncausal Systems:
a. Causal - output of the system at any time n depends only on present and past inputs
*x(n), x(n-1), x(n-2)
b. NonCausal - depends on future inputs
*x(n+1), x(n+2), x(n+3)
*physically unrealizable!
x(n) - x(n-1)

Causal
Causal
Causal
Noncausal
*Advanced
Non Causal
*If negative, it depends on its positive value.
Non Causal
*If positive, it depends on the value of its double.
Non Causal
*If negative, it depends on the value of its positive.

Sample Problems:
a.

Causal

b.

Non Causal

c.

Causal

d.

Non Causal

e.

Causal

f.

Causal

g.

Causal

h.

Causal

i.

Non Causal

j.

Non Causal

k.

Causal

l.

Non Causal

m.

Causal

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Causal

5. Stability versus Unstable Systems:


*Arbitrary relaxed system - bounded input-bounded output (BIBO) stable if and only if every bounded input produces a bounded output.
a. Stable - bounded input produced bounded output --> value approaches zero as time goes by
b. Unstable - value approaches infinity or nonzero.
*Static, causal, stable

Sample Problems:
a.

Stable
*Less than 1, kapag n = infinity

b.

Unstable
*Summation

c.

Stable
*Less than 1, kapag n = infinity

d.

Stable

e.

Stable

f.

Stable

g.

Stable

h.

Stable

i.

Unstable

j.

Stable

k.

Stable

l.

Stable

m.

Stable

Stable

a.

Stable
*Less than 1, kapag n = infinity

b.

Unstable
*Summation

c.

Stable
*Less than 1, kapag n = infinity

d.

Stable

e.

Stable

f.

Stable

g.

Stable

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g.

Stable

h.

Stable

i.

Unstable

j.

Stable

k.

Stable

l.

Stable

m.

Stable

Stable

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Linear Time Invariant (LTI) Systems (Chapter 2.7 - JS


Chitode)
*Majority of the discrete time systems in practice are linear and shift invariant.
*Discrete time signal can be expressed in the form of weighted impulses or weighted unit samples.
*Linear Convolution: powerful analytical tool for studying LTI systems
*Linearity property states that the output due to a linear combination of inputs is the same as the sum of
outputs due to individual inputs. (Distributive Property)
*Scaling property states that if:
*Shift Invariance = if the excitation of the shift invariant system is delayed, then its response is also delayed by
the same amount.
2.7.1 Discrete Time Signal as Weighted Impulses

*k = the range of the finite values.


*x(k) = amplitude of the unit sample
* (n-k) = location of the unit sample
2.7.2 Linear Convolution
*Linear Convolution - is a very powerful technique used for the analysis of LTI systems.

*If x(n) is applied as an input:

*It can be expanded as:

*Since LTI systems are linear, the transformation T can be distributed:

*Since LTI systems are linear, the scaling property can also be applied, where the magnitudes can be treated as
"coefficients".

*h(n) = unit sample response / impulse response of the system


Since LTI systems are shift invariant,

Substituting the response of the unit sample, we can derive the response of an LTI system, which is the linear
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Substituting the response of the unit sample, we can derive the response of an LTI system, which is the linear
convolution::

Operations in Computation of Convolution:

Sample Problem 1:
*Find the convolution of:
x(n) = [1,1,1,1]
h(n) = [2,2]
Tabular Method:
1

-1

x(k)

h(k)

h(-k)

h(-k+1)

h(-k+2)

h(-k+3)

h(-k+4)

1. Folding: h(-k)
2. Shifting h(-k+1), h(-k+2), h(-k+3) and h(-k+4)
3. Multiplication and Summation:

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Convolution: y(n) = [2,4,4,4,2]


*If there are M number of samples in x(n) and N number of samples in h(n); then the maximum number of
samples in y(n) is equal to M+N-1.
Basic Convolution Formula Method

2.7.3 Properties of Convolution:


1. Commutative Property of Convolution

Let m = n - k, hence k=n-m. Substituting:

Note that m is only a dummy index and it can be replaced by any character. Let m = k, hence

2. Associative Property of Convolution


Left side:

Right side:

*So if there are L numbers of LTI systems connected in cascade with unit sample responses of systems,
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*So if there are L numbers of LTI systems connected in cascade with unit sample responses of systems,
h(n) = h1(n) * h2(n) *h3(n) ** hL(n)
3. Distributive Property of Convolution:

*It is possible to decompose a complex system into parallel connection of small and simplified systems.
h(n) = h1(n) + h2(n) + h3(n) ++ hL(n)
2.7.4 Causality of LTI Systems:

2.7.5 Stability of LTI Systems:


*Stable System- if it produces bounded output for every bounded input.
*Stable LTI System - if its unit sample response is absolutely summable.

*Steps for Convolution by Multiplication:


1. Convert x(n) and h(n) to Sequence form.
2. Multiply x(n) and h(n). Magkaharap sila sa left. Start multiplying from the right.
3. The zeroth position will be determined by adding how many n distances the first terms of x(n) and h(n) from
zero.

*Steps for Checking if the convolution is correct:


1. Get the sum of all x(n)'s terms.
2. Get the sum of all h(n)'s terms.
3. Multiply 1 and 2.

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Difference Equations (Chapter 2.4 - Chitode)


*Difference Equations: an efficient way to implement discrete time systems

Type Types of Systems depending upon the number of Samples:


Types

Samples

Implementation

Memory

Others:

1. Finite Impulse Response (FIR)

Convolution

Requires M-1 past input


memory

*Becomes inefficient when M


is large

2. Infinite Impulse Response (IIR)

Difference Equations

Requires infinite or very


large memory for past
inputs

*not possibly impractical in


convolution since it requires
infinite memory

Equations:
1. Finite Impulse Response (FIR)

2. Infinite Impulse Response (IIR) Systems:

Classifications

Depends on:

Feedback

Applicability

Others

1. Nonrecursive Systems

Inputs (Past and


present)

None

FIR Systems only but


needs large memory

Practically impossible on IIR systems that


require infinite memory

Inputs (Past and


present) and Outputs
(Past)

Yes

Both FIR and IIR

efficient in terms of memory


requirement and computations

2. Recursive Systems

*Past sample outputs are called feedback


Equations and Diagrams:
1. Nonrecursive Systems:

2. Recursive System

2.8.5 Representation of Discrete Time Systems via Difference Equation:


*Constant coefficient difference equations:

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*Constant coefficient difference equations:

Initial Condition
1.

y(-1) is 0

2. y(-1) is not 0

System

Output

Initially Relaxed / Zero State of the System Zero State Response or Forced Response
Not Initially Relaxed

Zero Input Response or Natural Response

Example:

Where N = order of the difference equations


a and b = constant coefficients
x = input signal
y = output signal

Characteristics of Difference Equations:


1. Linear:
a. Total response y(n) = sum of zero input response YZI and
b. Zero state response YZS, and zero input response YZI and YZS should be linear
2. Shift Invariance - coefficients a and b are constants, hence won't affect time-invariance
3. Stability - should be BIBO

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Notation

YZS
YZI

Difference Equations (Chapter 2.4 - Proakis)


*FIR can be implemented through the convolution summation.
*IIR is practically impossible to implement in convolution summation, since it requires an infinite number of memory
locations, multiplications and additions.
2.4.1 Recursive and Nonrecursive Discrete-Time Systems
*Convolution summation formula: expresses the output of an LTI system only in terms of the input signal
Example 1: How to convert the expression into a recursive expression

Difference between Recursive and Nonrecursive


1. Recursive has feedback loop with delay element.
2. In recursive systems, to get y(n0), one must get ALL the values of the previous outputs first. In nonrecursive system,
y(n0) can be computed immediately.
3. The output of a recursive system has to be computed in order, while the output of nonrecursive system can be in any
order.
2.4.2 Linear Time-Invariant Systems Characterized by Constant-Coefficient Difference Equations

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Implementation of Discrete-Time Systems


*In practice, system design and implementation are usually treated jointly rather than separately.
2.5.1 Structures for the Realization of Linear Time-Invariant Systems
*Order of the system depends only on the output variable y!
*y(n-N) where N is the system order.
*Direct Form I Structure = separate delays (memory) for both the input and output signals
*Direct Form II Structure (also called canonic form) = simplified form 1, which is more efficient in terms of memory
requirements.
Example 1:

Example 2:

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*Direct Form 1 requires M+N delays and M+N+1 multiplication

Direct Form 1 Realization: M+N delays and M+N+1 multiplications

Direct Form II: M or N delays, depending on which is bigger and M+N+1 multiplications

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Example 3:

2.5.2 Recursive and Nonrecursive Realizations of FIR Systems


*FIR Systems can be implemented recursively or nonrecursively
*Weighted Moving Average System - FIR System

*FIR and IIR - general characteristics that distinguish a type of linear time-invariant system
*Recursive and non-recursive - descriptions of the structures for realizing or implementing the system
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*Recursive and non-recursive - descriptions of the structures for realizing or implementing the system

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Solution of Linear Constant-Coefficient Difference


Direct Method

- Determining explicit expression out of an LCCDE equation

Indirect Method - Through Z-transform

Direct Method:

Where yh(n) = homogenous or complementary solution


Yp(n) = particular solution

Homogenous Difference Equation:

Where:

Substituting it to the above equation:

Expanding it:

Where the polynomials within the parenthesis is the characteristic polynomial of the system.
*N is also the number of roots.
*Complex-valued roots occur in complex-conjugate pairs.
After getting the roots:
Homogenous Solution or the Zero-Input Response:

Steps
1. Let x(n) = 0, since it is a zero-input response we're getting and substitute.
2. Let y(n) = n and substitute.
3. Get the values of Lambda and substitute on the homogenous equation:

4. To get the values of C variables, get y(0) and y(1) or more on both the above equation and the original equation y(n). Use systems
of linear equations to get the C variables.
5. Substitute the C variables and make sure to put the range of N.
6. Substitute any initial conditions.
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6. Substitute any initial conditions.


Reminders:
1. Don't forget the range for n!

Particular Solution

Steps:
1. Check the form of the input signal. Let y(n) be equal to that assumed form multiplied by a constant.
2. Substitute the said y(n) equivalency to the original equation.
3. Assume a value of N that would not cancel any terms in the equation.
4. Get the value of K.
5. Substitute it to the assumed form.

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The Z-transform and Its Application to the Analysis of LTI


Systems (Chapter 3 - Proakis)
Z-Transform

Analysis of discrete time signals

Laplace Transform Analysis of continuous time signals.


*Convolution of two time-domain signals = multiplication of their corresponding z-transforms
3.1.1 The Direct Z-Transform
*Transforms the time-domain signal x(n) into its complex-plan representation X(z).
*A more compact alternative representation of signal

*Z is a complex variable
*Z-n value of the signal at time n.
*Since the z-transform is an infinite power series, it exists only for those values of z for which this series
converges.
*Region of Convergence - set of all values of z for which X(z) attains a finite value.
*ROC of a finite-duration signal is the entire z plane except 0 and / or infinity
*ROC of an infinite-duration, two sided signal, it is a ring (annular region) in the z-plane.
*Laging absolute value yung z!
Causal Signal

|z|>|a|
*Exterior of a circle of a certain radius.

Anticausal Signal

|z|<|a|
*Interior of a circle of a certain radius a.

*This implies that a closed-form expression for the z-transform does not uniquely specify the signal in
the time domain.
*The ambiguity can be resolved only if in addition to the closed-form expression, the ROC is specified.
*A discrete-time signal x(n) is uniquely determined by its z-transform X(z) and the region of
convergence X(z).

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A/D Conversion Process (2.10 - Chitode)


*Most of the signals are analog in nature when they are generated from the primary source.

2.10.1 Sampling
Sampling - Conversion of a raw continuous time signal xa(t) to discrete time signal x(n) by taking
samples at regular sampling time intervals (T).

*Sampler is normally a switch and it chops off the incoming analog signal at a sampling rate of Fs.
*Fs is also called sampling frequency (or samples per second).

2.10.2 Frequency Relationships


*Continuous time frequencies (F, ) to Discrete time frequencies (f,)

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*If the signal frequency lies in the range -Fs/2 to Fs/2, it can be represented by discrete time signal!

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Inversion of the Z-Transform (3.4 - Proakis)


*integral is a contour integral over a closed path C that encloses the origin and lies within the region of
convergence of X(z)
Three Methods:
1. Direct evaluation by contour integration.
2. Expansion into a series of terms, in the variables z and z-1.
3. Partial-fraction expansion and table lookup.
3.4.1 The Inverse Z-Transform by Contour Integration

3.4.1 Inverse Z-transform Using Partial Fraction Expansion


In this method, the function X(z) is expressed as:
*Where all the a are constants
*Where all X are standard z-transforms.

Hence, by linearity property, to get the z-transform, we can obtained a sequence x(n) such as:

Applicable for z-transforms of rational nature (proper form):

*Where a0 = 1 and M<N


a0 has to be made equal to 1:

Multiplying both numerator and denominator by ZN:

Extract z and transfer to the left side:

The denominator can then be written as this and can be factored like this:

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Equation can then be written as:

And can be expressed as partial fraction like this:

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Inversion of the Z-Transform (3.4 - Proakis)


*integral is a contour integral over a closed path C that encloses the origin and lies within the region of
convergence of X(z)
Three Methods:
1. Direct evaluation by contour integration.
2. Expansion into a series of terms, in the variables z and z-1.
3. Partial-fraction expansion and table lookup.
3.4.1 The Inverse Z-Transform by Contour Integration

3.4.1 Inverse Z-transform Using Partial Fraction Expansion


In this method, the function X(z) is expressed as:
*Where all the a are constants
*Where all X are standard z-transforms.

Hence, by linearity property, to get the z-transform, we can obtained a sequence x(n) such as:

Applicable for z-transforms of rational nature (proper form):

*Where a0 = 1 and M<N


a0 has to be made equal to 1:

Multiplying both numerator and denominator by ZN:

Extract z and transfer to the left side:

The denominator can then be written as this and can be factored like this:

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Equation can then be written as:

And can be expressed as partial fraction like this:

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Laboratory #4 (AD and DA Conversion)


*Analog-to-digital signal conversion - converting a continuous time and amplitude signal into discretetime and amplitude values.
1. Sampling
2. Quantization
3. Coding

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Partial Fraction
1.
2.
3.
4.
5.

Convert first to proper fraction by long division.


Multiply both sides by ZN to eliminate negative exponents.
Factor out denominator at ihiwalay into two different fractions.
Kapag distinct poles: A, B
Kapag multiple-order poles:

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Z-Transform Tables
Wednesday, October 07, 2015

10:50 AM

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