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UNIT-1 Signals and Systems

The document outlines the fundamentals of Digital Signal Processing (DSP), covering topics such as signals and systems, discrete-time signals, sampling, quantization, and system properties. It emphasizes the importance of understanding DSP principles for applications in various fields, including telecommunications and multimedia. The course objectives include knowledge of signal representation, transformation techniques, filter design, and the effects of aliasing.
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0% found this document useful (0 votes)
6 views51 pages

UNIT-1 Signals and Systems

The document outlines the fundamentals of Digital Signal Processing (DSP), covering topics such as signals and systems, discrete-time signals, sampling, quantization, and system properties. It emphasizes the importance of understanding DSP principles for applications in various fields, including telecommunications and multimedia. The course objectives include knowledge of signal representation, transformation techniques, filter design, and the effects of aliasing.
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as PDF, TXT or read online on Scribd
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DIGITAL SIGNAL PROCESSING

Course Code: 21EEC302T


UNIT – 1

Signals and Systems


UNIT-1 Signals and Systems
Overview of signals and systems, Standard Discrete time signals-
Classifications of Discrete time signals, Sampling and Sampling
Theorem, Quantization and Coding- Aliasing, Effects of Aliasing -system
properties (linearity, time-invariance, memory, causality, BIBO stability)
- LTI systems described by linear constant coefficient difference
equations (LCCDE) - impulse response and convolution-Simulation on
continuous time signals, Simulation on discrete time signals, systems.
Overview Of Digital Signal Processing
• Digital Signal Processing is the branch of engineering that has enabled unprecedented levels of
interpersonal communication and of on-demand entertainment. In this course we will cover the
basics of Digital Signal Processing.
• We will concentrate on the basic mathematical formulations, rather than in-depth
implementation details. We will cover the breadth of topics, beginning with the basics of
signals and their representations, the theory of sampling, difference equations, important
transform representations like Fourier transform, Z transform, Analog and Digital filter
design and architectures, addressing modes of digital signal processor.
• A thorough understanding of digital signal processing fundamentals and techniques is
essential for anyone whose work is concerned with signal processing applications. By
reworking the principles of electronics, telecommunication and computer science into a unifying
paradigm, DSP is the heart of the digital revolution that brought us CDs, DVDs, MP3
players, mobile phones, multimedia, security, health care and countless other devices.
Course Objectives
To impart knowledge about the following topics:
• Signals and systems & their mathematical representation.
• Discrete time systems.
• Transformation techniques & their computation.
• Filters and their design for digital implementation.
• Programmable digital signal processor & quantization effects.
Course Outcomes
• Identify the continuous and discreate-time signals and systems.
• Apply the concept of DFT for transforming time domain signal to frequency
domain signal
• Design the digital IIR and FIR filters
• Analyze the concept of multirate signal processing
Overview of signals and systems

Signal
• A signal is defined as any physical quantity that varies with time,
space or any other independent variable or variables.
Eg : x(t) = 2t, x(t) = 2t2

System
• A system is defined as a physical device that performs an
operation on a signal.
Eg: Analog signal processing system, Digital Signal Processing
System.
Analog Signal Processing System

Analog Input Signal Analog Output Signal


Analog Signal
Processor
Digital Signal Processing System

Anti Sample
Aliasing + A/D DSP D/A Reconstruction
Converter Converter Filter
Filter Hold
Classification of Discrete-Time Signal

1.Energy and Power Signals


2. 2. Periodic and Aperiodic Signals
3.3. Symmetric (Even) and Antisymmetric (Odd) Signals
4. 4. Causal and Non Causal Signals
5.5. Deterministic and Random Signals
1. Energy and Power Signals
The Energy E of a signal x(n) is defined as

The energy of a signal can be finite or infinite. If E is finite (i.e., 0 <


E < ∞), then x(n) is called an energy signal.
The average power of a discrete-time signal x(n) is defined as

If E is finite, P = 0. On the other hand, if E is infinite, the average


power P may be either finite or infinite. If P is finite (and nonzero), the
signal is called a power signal.
2. Periodic and Aperiodic Signals
A signal x(n) is periodic with period N(N > 0) if and only if

The smallest value of N for which above equation holds is called


the fundamental period. If there is no value of N that satisfies above
equation , the signal is called nonperiodic or aperiodic.
2. Periodic and Aperiodic Signals
• A discrete-time signal x(n) is said to be periodic if it satisfies
the condition x(n) = x(n + N) for all integers n.
• The smallest value of N which satisfies the above condition is
known as fundamental period
• If the above condition is not satisfied even for one value of n,
then the discrete-time signal is aperiodic. Sometimes
aperiodic signals are said to have a period equal to infinity.
• The angular frequency is given by
3. Even and Odd Signals
• Any signal x(n) can be expressed as sum of even and odd components. xe(n) is
even components and xo(n) is odd components of the signal.
• A discrete-time signal x(n) is said to be an even (symmetric) signal if it satisfies the
condition:
x(n) = x(–n) for all n
• Even signals are symmetrical about the vertical axis or time origin. Hence they are
also called symmetric signals: cosine sequence is an example of an even signal. An
even signal is identical to its reflection about the origin.
3. Even and Odd Signals
• A discrete-time signal x(n) is said to be an odd (anti-
symmetric) signal if it satisfies the condition:
x(–n) = –x(n) for all n
• Odd signals are anti-symmetrical about the vertical axis.
Hence they are called antisymmetric signals. Sinusoidal
sequence is an example of an odd signal.
3. Even and Odd Signals
4. Causal and Non-causal Signals
• A discrete-time signal x(n) is said to be causal if x(n) = 0 for n < 0,
otherwise the signal is non-causal. A discrete-time signal
• x(n) is said to be anti-causal if x(n) = 0 for n > 0.
• A causal signal does not exist for negative time and an anti-
causal signal does not exist for positive time. A signal which
exists in positive as well as negative time is called a non-
casual signal.
• u(n) is a causal signal and u(– n) an anti-causal signal, whereas
• x(n) = 1 for – 2 ≤ n ≤ 3 is a non-causal signal.
5. Deterministic and Random Signals
A signal exhibiting no uncertainty of its magnitude and phase at any given instant of
time is called deterministic signal. A deterministic signal can be completely
represented by mathematical equation at any time and its nature and amplitude at any
time can be predicted.
Eg: Sinusoidal sequence, Exponential sequence, Ramp Sequence.

A signal characterized by uncertainty about its occurrence is called non-deterministic


or random signal. A random signal cannot be represented by mathematical equation.
The behavior of the signal is probabilistic in nature and can be analyzed only
stochastically. The pattern of such a signal is quite irregular. Its amplitude and phase at
any time cannot be predicted in advance. Eg: Thermal noise
Standard Discrete Time Signals

1. Digital impulse signal


2. Unit step signal
3. Unit ramp signal
4. Exponential signal
Sampling, Quantization, and Encoding
Sampling Theorem (Nyquist-Shannon Sampling Theorem)
A Band Limited continuous time signal with the maximum frequency fm can be fully
recovered from its samples provided that sampling frequency fm is greater than or equal to two
times the maximum frequency fm
(Fs) or F ≥ 2fm
where Fs is the sampling frequency (or sampling rate)
Fmax is the highest frequency component present in the baseband signal.
The time interval T between successive samples is called the sampling period or sample
interval and its reciprocal 1/T = Fs is called the sampling rate (samples per second)
or the sampling frequency (hertz).
i.e., sampling frequency must be at least twice the higher frequency in the signal.

Nyquist Rate which denotes the lowest rate at which a signal must be sampled in order to
prevent information loss. The highest frequency component of the baseband signal is double
what the Nyquist Rate is
Sampling Theorem (Nyquist-Shannon Sampling Theorem)
Sampling of Analog Signals
Sampling Theorem (Nyquist-Shannon Sampling Theorem)
Sampling Theorem (Nyquist-Shannon Sampling Theorem)
Quantization
Quantization Definition : The process of converting a continuous analog signal into a digital signal by
expressing each sample value as a finite (instead of an infinite) number of digits is called
quantization. It is split into quantized fixed intervals, each of which corresponds to a distinct digital code or
level
Quantization Error : The error introduced in representing the continuous- valued signal by a finite set of
discrete value levels is called quantization error or quantization noise.
Then the quantization error is a sequence eq(n) defined as the difference between the quantized
value and the actual sample value. Thus
eq(n) = xq(n) − x(n)
eq(n)- Quantization Error
xq(n) - quantized value
x(n)- actual sample value
Quantization
Coding
The process of converting an analog signal to a digital signal
continues with encoding after the analog signal has been
sampled and quantized. Each quantized value is represented
using a binary code in this manner. The resolution of binary
encoding, which employs combinations of 0s and 1s,
depends on the amount of bits utilized in this representation
Aliasing
• Aliasing is a phenomenon that occurs in digital signal processing (DSP) when a
continuous signal is sampled at a rate that is insufficient to capture the changes in
the signal accurately. It results in different signals becoming indistinguishable (or
aliases of each other) when sampled. This can cause significant distortion and loss of
information in the reconstructed signal.
• A low pass anti-aliasing filter is employed, before the sampler, to
eliminate the high frequency components, which are unwanted.
• The signal which is sampled after filtering, is sampled at a rate slightly higher than
the Nyquist rate.
Anti-Aliasing Filter
In practice, communication signals have frequency spectra low
consisting of frequency components as well as high-
frequency noise components. If we select sampling frequency F, all signals
with frequency higher than Ωs/2 appears as signals of frequencies between 0
and Ωs/2 due to aliasing effect. To avoid aliasing we can choose very
high sampling frequency. But sampling at very high frequencies
introduces numerical errors. Therefore, to avoid aliasing errors caused by
the undesired high frequency signals, an analog lowpass filter , called anti-
aliasing filter is used prior to sampler to filter high frequency components
before the signal is sampled.

Anti Sample
Aliasing + A/D DSP D/A Reconstruction
Converter Converter Filter
Filter Hold
Effects of Aliasing
There are a few effects of aliasing and why it should be prevented:
•Signal Accuracy: Aliasing makes signal become distorted which can cause unwanted problems in any signal. This
can be a major problem in Audio, which can cause audio instruments to sound distorted and also in Video, which
can cause sharp/pixelated or jagged edges in pictures.
•Distortion: The most noticeable effect of aliasing is distortion. The reconstructed signal does not accurately
represent the original signal, as high-frequency components are incorrectly mapped to lower frequencies.
•Information Loss: Important information contained in the high-frequency components of the signal is lost due to
aliasing, making the reconstruction of the original signal impossible.
•Misinterpretation: Aliasing can cause significant errors in signal interpretation, especially in applications requiring
precise measurements, such as biomedical signal processing or audio engineering.
•Reconstruction of Signal: Due to aliasing, it may become impossible to perfectly reconstruct orignal signal from
its sample because of data loss. Hence aliasing can make reconstruction of signals hard.
•Signal processing: Aliasing can make signal processing complicated by producing unwanted noise.
•Poor signal quality: Aliasing can negatively impact any signals quality and cause distortion and corruption of the
signal.
CAUSALITY- CASUAL SYSTEMS
A system is causal if the output at any time depends only on the current and past inputs, not on
future inputs. This is essential for real-time processing.
Mathematical Definition:

a) y(t) = x(t) if we substitute t = 3, the result will show for that instant of time only. Therefore, as it has no
dependence on future value, we can call it a Causal system.
b) y(t) = x(t-1) Here, the system depends on past values. For instance if we substitute t = 3, the expression
will reduce to x , which is a past value against our input. At no instance, it depends upon future values.
Therefore, this system is also a causal system.
c) y(t) = x(t) + x(t+1) In this case, the system has two parts. The part x , as we have discussed earlier,
depends only upon the present values. So, there is no issue with it. However, if we take the case of x , it
clearly depends on the future values because if we put t = 1, the expression will reduce to x which is
future value. Therefore, it is not causal. y(t) = x(t)
CAUSALITY- Non-Causal Systems
A non-causal system is just opposite to that of causal system. If a system depends upon the future
values of the input at any instant of the time then the system is said to be non-causal system.

a) y(t)=x(t+1)
We have already discussed this system in causal system too. For any input, it will reduce the
system to its future value. For instance, if we put t = 2, it will reduce to x(3)
, which is a future value. Therefore, the system is Non-Causal.

b) y(t)=x(t)+x(t+2)

In this case, x(t) is purely a present value dependent function. We have already discussed that
x(t+2) function is future dependent because for t = 3 it will give values for x(5)
Therefore, it is Non-causal.

c) y(t)=x(t−1)+x(t)

In this system, it depends upon the present and past values of the given input. Whatever values we
substitute, it will never show any future dependency. Clearly, it is not a non-causal system; rather it
is a Causal system.
CAUSALITY - Anti-Causal Systems
An anti-causal system is just a little bit modified version of a non-causal system. The system depends upon the future
values of the input only. It has no dependency either on present or on the past values.

Examples
Find out whether the following systems are anti-causal.

a) y(t)=x(t)+x(t−1)

The system has two sub-functions. One sub function x(t+1) depends on the future value of the input but another
sub-function x(t) depends only on the present. As the system is dependent on the present value also in addition to
future value, this system is not anti-causal.

b) y(t)=x(t+3)

If we analyze the above system, we can see that the system depends only on the future values of the system i.e. if
we put t = 0, it will reduce to x(3), which is a future value. This system is a perfect example of anti-causal system.
LINEARITY- linear system
A linear system follows the laws of superposition. This law is necessary
and sufficient condition to prove the linearity of the system. Apart from
this, the system is a combination of two types of laws −
•Law of additivity
•Law of homogeneity
The conditions are −
The output should be zero for zero input.
There should not be any non-linear operator present in the system.
Examples of non-linear operators −
a)Trigonometric operators- Sin, Cos, Tan, Cot, Sec, Cosec etc.
b Exponential, logarithmic, modulus, square, Cube etc.
c)sai/p, Sinc i/p, Sqn i/p etc.
Either input x or output y should not have these non-linear operators.
Examples
a) y(t)=x(t)+3

This system is not a linear system because it violates the first condition. If we put input
as zero, making xt= 0, then the output is not zero.

b) y(t)=sintx(t)

In this system, if we give input as zero, the output will become zero. Hence, the first
condition is clearly satisfied. Again, there is no non-linear operator that has been applied
on xt . Hence, second condition is also satisfied. Therefore, the system is a linear
system.

c) y(t)=sin(x(t))

In the above system, first condition is satisfied because if we put xt= 0, the output will
also be sin0= 0. However, the second condition is not satisfied, as there is a non-linear
operator which operates xt. Hence, the system is not linear.
LINEARITY- Non-linear system
The systems, which are not linear are non-linear systems. Clearly, all the conditions, which are
being violated in the linear systems, should be satisfied in this case.

Conditions
• The output should not be zero when input applied is zero.

• Any non-linear operator can be applied on the either input or on the output to make the system
non-linear.
LINEARITY- Non-linear system

Examples
To find out whether the given systems are linear or non-linear.

a) y(t)=ex(t)

In the above system, the first condition is satisfied because if we make the input zero, the output
is 1. In addition, exponential non-linear operator is applied to the input. Clearly, it is a case of
Non-Linear system.

b) y(t)=x(t+1)+x(t−1)

The above type of system deals with both past and future values. However, if we will make its
input zero, then none of its values exists. Therefore, we can say if the input is zero, then the time
scaled and time shifted version of input will also be zero, which violates our first condition. Again,
there is no non-linear operator present. Therefore, second condition is also violated. Clearly, this
system is not a non-linear system; rather it is a linear system.
Time-Invariant Systems
For a time-invariant system, the output and input should be delayed by some time unit.
Any delay provided in the input must be reflected in the output for a time invariant
system.
Time-Invariant Systems

a) 𝑦(𝑇)=𝑥(2𝑇)
If the above expression, it is first passed through the system and then through the
time delay as shown in the upper part of the figure; then the output will
become 𝑥(2𝑇−2𝑡). Now, the same expression is passed through a time delay first and
then through the system as shown in the lower part of the figure. The output will
become 𝑥(2𝑇−𝑡).
Hence, the system is not a time-invariant system.
b) 𝑦(𝑇)=sin[𝑥(𝑇)]
If the signal is first passed through the system and then through the time delay
process, the output be sin𝑥(𝑇−𝑡). Similarly, if the system is passed through the time
delay first then through the system then output will be sin𝑥(𝑇−𝑡). We can see clearly
that both the outputs are same. Hence, the system is time invariant.
Stable Systems – BIBO

A stable system satisfies the BIBO (bounded input for bounded


output) condition. Here, bounded means finite in amplitude.
For a stable system, output should be bounded or finite, for
finite or bounded input, at every instant of time.

Unstable systems do not satisfy the BIBO conditions.


Therefore, for a bounded input, we cannot expect a bounded
output in case of unstable systems.
Example 1 − Check whether y(t)=x∗(t)𝑦(𝑡)=𝑥∗(𝑡) is linear or non-linear.
Solution − The function represents the conjugate of input. It can be verified by either first
law of homogeneity and law of additivity or by the two rules. However, verifying through
rules is lot easier, so we will go by that.
If the input to the system is zero, the output also tends to zero. Therefore, our first
condition is satisfied. There is no non-linear operator used either at the input nor the
output. Therefore, the system is Linear.
Example 2 − Check whether the system is linear or non linear

Solution − Clearly, we can see that when time becomes less than or equal to zero the input becomes zero.
So, we can say that at zero input the output is also zero and our first condition is satisfied.
Again, there is no non-linear operator used at the input nor at the output. Therefore, the system is Linear.
Example 3 − Check whether 𝑦(𝑡)=sin𝑡.𝑥(𝑡) is stable or not.
Solution − Suppose, we have taken the value of xt𝑡 as 3. Here, sine function has been multiplied with it and
maximum and minimum value of sine function varies between -1 to +1.
Therefore, the maximum and minimum value of the whole function will also vary between -3 and +3. Thus,
the system is stable because here we are getting a bounded input for a bounded output.
MEMORY
A system has memory if its output depends on past or future inputs. A memoryless
system's output depends only on the current input.

Mathematical Definition:
A system S is memoryless if the output y(t) at any time t depends only on the input x(t)
at that same time:
y(t)=S[x(t)]
Linear Constant Coefficient Difference Equations (LCCDE)

A linear constant-coefficient difference equation (LCCDE) serves as a way to express just this
relationship in a discrete-time system. Writing the sequence of inputs and outputs, which represent
the characteristics of the LTI system, as a difference equation help in understanding and
manipulating a system.

Definition: Difference Equation

An equation that shows the relationship between consecutive values of a sequence and the
differences among them. They are often rearranged as a recursive formula so that a systems output
can be computed from the input signal and past outputs.
Example :
y[n]+7y[n−1]+2y[n−2]=x[n]−4x[n−1]
Linear Constant Coefficient Difference Equations (LCCDE)

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