UNIT I – INTRODUCTION

Classification of systems: Continuous, discrete, linear, causal, stability, dynamic,

recursive, time variance; classification of signals: continuous and discrete, energy
and power; mathematical representation of signals; spectral density; sampling
techniques, quantization, quantization error, Nyquist rate, aliasing effect.

SIGNAL

A SIGNAL is defined as any physical quantity that changes with time, distance, speed, position,
pressure, temperature or some other quantity. A SIGNAL is physical quantity that consists of
many sinusoidal of different amplitudes and frequencies.

Example:

x(t) = 10t

X(t) = 5x2+20xy+30y

SYSTEM

A System is a physical device that performs an operations or processing on a signal. Ex Filter or

Amplifier.

CLASSIFICATION OF SIGNAL PROCESSING

1) ASP (Analog signal Processing) : If the input signal given to the system is
analog then system does analog signal processing. Ex Resistor, capacitor or
Inductor, OP-Amps etc.
 2) DSP (Digital signal Processing): If the input signal given to the system is digital then
system does digital signal processing. Ex Digital Computer, Digital Logic Circuits etc.
The devices called as ADC (analog to digital Converter) converts Analog signal into
digital and DAC (Digital to Analog Converter) does vice-versa.

Most of the signals generated are analog in nature. Hence these signals are converted to
digital form by the analog to digital converter. Thus AD Converter generates an array of
samples and gives it to the digital signal processor. This array of samples or sequence of
samples is the digital equivalent of input analog signal. The DSP performs signal
processing operations like filtering, multiplication, transformation or amplification etc
operations over these digital signals. The digital output signal from the DSP is given to
the DAC.

ADVANTAGES OF DSP OVER ASP:

1. Physical size of analog systems is quite large while digital processors are more compact and
light in Weight.
2. Analog systems are less accurate because of component tolerance ex R, L, C and active
components.
Digital components are less sensitive to the environmental changes, noise and disturbances.
3. Digital system is most flexible as software programs & control programs can be easily
modified.
4. Digital signal can be stores on digital hard disk, floppy disk or magnetic tapes. Hence becomes
Transportable. Thus easy and lasting storage capacity.
5. Digital processing can be done offline.
6. Mathematical signal processing algorithm can be routinely implemented on digital signal
processing systems. Digital controllers are capable of performing complex computation with
constant accuracy at high speed.
7. Digital signal processing systems are upgradeable since that are software controlled.
8. Possibility of sharing DSP processor between several tasks.
9. The cost of microprocessors, controllers and DSP processors are continuously going down.
For some complex control functions, it is not practically feasible to construct analog controllers.
10. Single chip microprocessors, controllers and DSP processors are more versatile and
powerful.

Disadvantages of DSP over ASP

1. Additional complexity (A/D & D/A Converters)
2. Limit in frequency. High speed AD converters are difficult to achieve in practice. In high
frequency applications DSP are not preferred.
CLASSIFICATION OF SIGNALS
1. Single channel and Multi-channel signals
2. Single dimensional and Multi-dimensional signals
3. Continuous time and Discrete time signals.
4. Continuous valued and discrete valued signals.
5. Analog and digital signals.
6. Deterministic and Random signals
7. Periodic signal and Non-periodic signal
8. Symmetrical(even) and Anti-Symmetrical(odd) signal
9. Energy and Power signal

1. Single channel and Multi-channel signals

If signal is generated from single sensor or source it is called as single channel signal. If the
signals are generated from multiple sensors or multiple sources or multiple signals are generated
from same source called as Multi-channel signal. Example ECG signals. Multi-channel signal
will be the vector sum of signals generated from multiple sources.

2. Single Dimensional (1-D) and Multi-Dimensional signals (M-D)

If signal is a function of one independent variable it is called as single dimensional signal like
speech signal and if signal is function of M independent variables called as Multi - dimensional
signals. Gray scale level of image or Intensity at particular pixel on black and white TV is
examples of M-D signals.

3. Continuous time and Discrete time signals.

Continuous Time (CTS)

1. This signal can be defined at any time instance & they can take all values in the
continuous interval(a, b) where a can be -∞ & b can be ∞
2. These are described by differential equations.
3. This signal is denoted by x(t).
4. The speed control of a dc motor using a tacho generator feedback or Sine or
exponential waveforms.
 Discrete time (DTS)
1. This signal can be defined only at certain specific values of time. These time instance
need not be equidistant but in practice they are usually takes at equally spaced intervals.
2. These are described by difference equation.
3. These signals are denoted by x(n) or notation x(nT) can also be used.
4. Microprocessors and computer based systems uses discrete time signals.

4. Continuous valued and Discrete Valued signals.

Continuous Valued

1. If a signal takes on all possible values on a finite or infinite range, it is said to be continuous
valued signal.

2. Continuous Valued and continuous time signals are basically analog signals.
Discrete Valued

1. If signal takes values from a finite set of possible values, it is said to be discrete valued signal.
2. Discrete time signal with set of discrete amplitude are called digital signal.

5. Analog and digital signal

Analog signal

1. These are basically continuous time & continuous amplitude signals.

2. ECG signals, Speech signal, Television signal etc. All the signals generated from

Various sources in nature are analog.

Digital signal

1. These are basically discrete time signals & discrete amplitude signals. These signals are

basically obtained by sampling & quantization process.

2. All signal representation in computers and digital signal processors are digital.

Note: Digital signals (DISCRETE TIME & DISCRETE AMPLITUDE) are obtained by
sampling the ANALOG signal at discrete instants of time, obtaining DISCRETE TIME signals
and then by quantizing its values to a set of discrete values & thus generating DISCRETE
AMPLITUDE signals.

Sampling process takes place on x axis at regular intervals & quantization process takes place
along y axis. Quantization process is also called as rounding or truncating or approximation
process.

6. Deterministic and Random signals

Deterministic signals
1. Deterministic signals can be represented or described by a mathematical equation or lookup
table.
2. Deterministic signals are preferable because for analysis and processing of signals we can use
Mathematical model of the signal.
3. The value of the deterministic signal can be evaluated at time (past, present or future) without
Certainty.
4. Example Sine or exponential waveforms.

Random signals
1. Random signals that cannot be represented or described by a mathematical equation or lookup
table.
2. Not Preferable. The random signals can be described with the help of their statistical
properties.
3. The value of the random signal can not be evaluated at any instant of time.
4. Example Noise signal or Speech signal

7. Periodic signal and Non-Periodic signal

The signal x(n) is said to be periodic if x(n+N)= x(n) for all n where N is the fundamental period
of the signal. If the signal does not satisfy above property called as Non-Periodic signals.
Discrete time signal is periodic if its frequency can be expressed as a ratio of two integers. f=
k/N where k is integer constant.

DISCRETE TIME SIGNALS AND SYSTEM

There are three ways to represent discrete time signals.
1) Functional Representation
2) Tabular method of representation

3) Sequence Representation

1. STANDARD SIGNAL SEQUENCES

1) Unit sample signal (Unit impulse signal)

2) Unit step signal
3) Unit ramp signal
4) Exponential signal
5) Sinusoidal waveform

2. PROPERTIES OF DISCRETE TIME SIGNALS

1) Shifting: signal x(n) can be shifted in time. We can delay the sequence or advance the
sequence. This is done by replacing integer n by n-k where k is integer. If k is positive signal is
delayed in time by k samples (Arrow get shifted on left hand side) And if k is negative signal is
advanced in time k samples

2) Folding / Reflection : It is folding of signal about time origin n=0. In this case replace n by – n.
3) Addition : Given signals are x1(n) and x2(n), which produces output y(n) where y(n) = x1(n)
+ x2(n). Adder generates the output sequence which is the sum of input sequences.

4) Scaling: Amplitude scaling can be done by multiplying signal with some constant. Suppose
original signal is x(n). Then output signal is A x(n)

4) Multiplication : The product of two signals is defined as y(n) = x1(n) * x2(n).

3. SYMBOLS USED IN DISCRETE TIME SYSTEM

4. CLASSIFICATION OF DISCRETE TIME SYSTEMS

1. STATIC v/s DYNAMIC

It is very easy to find out that given system is static or dynamic. Just check that output of the system
solely depends upon present input only, not dependent upon past or future.

2) TIME INVARIANT v/s TIME VARIANT SYSTEMS

It is very easy to
find out that
given system is
Shift Invariant or
Shift Variant.
Suppose if the
system

produces output
y(n) by taking
input x(n)

x(n) -> y(n)

If we delay same input by k units x(n-k) and apply it to same systems, the system produces output y(nk)

x(n-k) -> y(n-k)

3) LINEAR v/s NON-LINEAR SYSTEMS

hence T [ a1 x1(n) + a2 x2(n) ] = T [ a1 x1(n) ] + T [ a2 x2(n) ] It is very easy to find out that
given system is Linear or Non-Linear.

Response to the system to the sum of signal = sum of individual responses of the system.
4) CAUSAL v/s NON CAUSAL SYSTEMS

CAUSAL
a) A System is causal if output of system at any time depends only past and present inputs.
b) In Causal systems the output is the function of x(n), x(n-1), x(n-2)….. and so on.
c) Example Real time DSP Systems

NON-CAUSAL (Causality Property)

a) A System is Non causal if output of system at any time depends on future inputs.
b) In Non-Causal System the output is the function of future inputs also. X(n+1) x(n+2) .. and so
on
c) Offline Systems
It is very easy to find out that given system is causal or non-causal. Just check that output of the
system depends upon present or past inputs only, not dependent upon future.

System [y(n)] Causal /Non-Causal

1 x(n) + x(n-3) Causal
2 X(n) Causal
3 X(n) + x(n+3) Non-Causal
4 2 x(n) Causal
5 X(2n) Non-Causal
6 X(n)+ x(n-2) +x(n+2) Non-Causal

5) STABLE v/s UNSTABLE SYSTEMS

STABLE
a) A System is BIBO stable if every bounded input produces a bounded output.
b) The input x(n) is said to bounded if there exists some finite number Mx such that
|x(n)| ≤ Mx < ∞
The output y(n) is said to bounded if there exists some finite number My such that
|y(n)| ≤ My < ∞

UNSTABLE (Stability Property)

a) A System is unstable if any bounded input produces a unbounded output.

STABILITY FOR LTI SYSTEM

It is very easy to find out that given system is stable or unstable. Just check that by providing
input signal check that output should not rise to ∞.

The condition for stability is given by

System [y(n)] Stable / Unstable

1 Cos [ x(n) ] Stable
2 x(-n+2) Stable
3 |x(n)| Stable
4 x(n) u(n) Stable
5 X(n) + n x(n+1) Unstable

BASIC BLOCK DIAGRAM OF A/D CONVERTER

SAMPLING THEOREM
It is the process of converting continuous time signal into a discrete time signal by taking
samples of the continuous time signal at discrete time instants.
X[n]= Xa(t) where t= nTs = n/Fs ….(1)
When sampling at a rate of fs samples/sec, if k is any positive or negative integer, we cannot
distinguish between the samples values of fa Hz and a sine wave of (fa+ kfs) Hz. Thus (fa + kfs)
wave is alias or image of a wave.

Thus Sampling Theorem states that if the highest frequency in an analog signal is Fmax and the
signal is sampled at the rate fs > 2Fmax then x(t) can be exactly recovered from its sample
values. This sampling rate is called Nyquist rate of sampling. The imaging or aliasing starts after
Fs/2 hence folding frequency is fs/2. If the frequency is less than or equal to 1/2 it will be
represented properly.

Thus the frequency 50 Hz, 90 Hz , 130 Hz … are alias of the frequency 10 Hz at the sampling
rate of 40 samples/sec

QUANTIZATION

The process of converting a discrete time continuous amplitude signal into a digital signal by
expressing each sample value as a finite number of digits is called quantization. The error
introduced in representing the continuous values signal by a finite set of discrete value levels is
called quantization error or quantization noise.

Quantization Step/Resolution : The difference between the two quantization levels is called
quantization step. It is given by Δ = XMax – xMin / L-1 where L indicates Number of
quantization levels.

CODING/ENCODING
Each quantization level is assigned a unique binary code. In the encoding operation, the
quantization sample value is converted to the binary equivalent of that quantization level. If 16
quantization levels are present, 4 bits are required. Thus bits required in the coder is the smallest
integer greater than or equal to Log2 L. i.e b= Log2 L Thus Sampling frequency is calculated as
fs=Bit rate / b.
ANTI-ALIASING FILTER
When processing the analog signal using DSP system, it is sampled at some rate depending upon
the bandwidth. For example if speech signal is to be processed the frequencies upon 3khz can be
used. Hence the sampling rate of 6khz can be used. But the speech signal also contains some
frequency components

more than 3khz. Hence a sampling rate of 6khz will introduce aliasing. Hence signal should be
band limited to avoid aliasing. The signal can be band limited by passing it through a filter (LPF)
which blocks or attenuates all the frequency components outside the specific bandwidth. Hence
called as Anti aliasing filter or pre-filter. (Block Diagram)

SAMPLE-AND-HOLD CIRCUIT:
The sampling of an analogue continuous-time signal is normally implemented using a device
called an analogue-to- digital converter (A/D). The continuous-time signal is first passed through
a device called a sample-and-hold (S/H) whose function is to measure the input signal value at
the clock instant and hold it fixed for a time interval long enough for the A/D operation to
complete. Analogue-to-digital conversion is potentially a slow operation, and a variation of the
input voltage during the conversion may disrupt the operation of the converter. The S/H prevents
such disruption by keeping the input voltage constant during the conversion. This is
schematically illustrated by Figure. After a continuous-time signal has been through the A/D
converter, the quantized output may differ from the input value. The maximum possible output
value after the quantization process could be up to half the quantization level q above or q below
the ideal output value. This deviation from the ideal output value is called the quantization error.
In order to reduce this effect, we increases the number of bits.

Q) Calculate Nyquist Rate for the analog signal x(t)

1) x(t)= 4 cos 50 Πt + 8 sin 300Πt –cos 100Πt Fn=300 Hz
2) x(t)= 2 cos 2000Πt+ 3 sin 6000Πt + 8 cos 12000Πt Fn=12KHz
3) x(t)= 4 cos 100Πt Fn=100 Hz
Q) The following four analog sinusoidal are sampled with the fs=40Hz. Find out corresponding
time signals and comment on them
X1(t)= cos 2Π(10)t
X2(t)= cos 2Π(50)t
X3(t)= cos 2Π(90)t
X4(t)= cos 2Π(130)t

Q) Signal x1(t)=10cos2Π(1000)t+ 5 cos2Π(5000)t. Determine Nyquist rate. If the signal is

sampled at 4khz will the signal be recovered from its samples.
Q) Signal x1(t)=3 cos 600Πt+ 2cos800Πt. The link is operated at 10000 bits/sec and each input
sample is quantized into 1024 different levels. Determine Nyquist rate, sampling frequency,
folding frequency & resolution.

DIFFERENCE BETWEEN FIR AND IIR

Discrete time systems have one more type of classification.

a) Recursive Systems
b) Non-Recursive Systems
UNIT-II
INTRODUCTION TO Z TRANSFORM
For analysis of continuous time LTI system Laplace transform is used. And for
analysis of discrete time LTI system z transform is used. Z transform is
mathematical tool used for conversion of time domain into frequency domain (z
domain) and is a function ofthe complex valued variable Z. The z transform of a
discrete time signal x(n) denoted by

X(z) and given as

Z transform is an infinite power series because summation index varies from -∞

to ∞. But it is useful for values of z for which sum is finite. The values of z for
which f (z) is finite and lie within the region called as “region of convergence
(ROC).

ADVANTAGES OF Z TRANSFORM
1. The DFT can be determined by evaluating z transform.

2. Z transform is widely used for analysis and synthesis of digital filter.

3. Z transform is used for linear filtering. z transform is also used for finding
Linear convolution, cross- correlation and auto-correlations of sequences.

4. In z transform user can characterize LTI system (stable/unstable,

causal/anti-causal) and its response to various signals by placements of pole and
zero plot.
ADVANTAGES OF ROC (REGION OF CONVERGENCE)
1. ROC is going to decide whether system is stable or unstable.

2. ROC decides the type of sequences causal or anti-causal.

3. ROC also decides finite or infinite duration sequences.

Z TRANSFORM PLOT

Fig show the plot of z transforms. The z transform has real and imaginary parts.
Thus a plot of imaginary part versus real part is called complex z-plane. The
radius of circle is 1 called as unit circle. This complex z plane is used to show
ROC, poles and zeros. Complex variable z is also expressed in polar form as Z=
rejω where r is radius of circle is given by |z| and ω is the frequency of the
sequence in radians and given by ∟z.
 Q) Determine z transform of following signals. Also draw ROC. i)

x(n)= {1,2,3,4,5} ii) x(n)={1,2,3,4,5,0,7}

Q) Determine z transform and ROC for x(n) = (-1/3)n u(n)

–(1/2)n u(-n-1). Q) Determine z transform and ROC for

x(n) = [ 3.(4n)–4(2n)] u(n).

Q) Determine z transform and ROC for x(n) = (1/2)n u(-n).

Q) Determine z transform and ROC for x(n) = (1/2)n {u(n) – u(n-10)}.

Q) Find linear convolution using z transform. X(n)={1,2,3} & h(n)={1,2}

PROPERTIES OF Z TRANSFORM (ZT)

1) Linearity

The linearity property states that if z

z Transform of linear combination of two or more signals is equal to the same linear
combination of z transform of individual signals.
2) Time shifting

The Time shifting property states that if z x(n)

Thus shifting the sequence circularly by „k samples is equivalent to multiplying its z transform by z –k

3) Scaling in z domain

This property states that if

Thus scaling in z transform is equivalent to multiplying by an in time domain.

4) Time reversal Property

The Time reversal property states that if z

It means that if the sequence is folded it is equivalent to replacing z by z-1 in z domain.

5) Differentiation in z domain

The Differentiation property states that if z