INTRODUCTION

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.

Ex

x(t) = 10t

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

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-AMP 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
(Arrow get shifted on right hand side)
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(n-
k)

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.

Sr No 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

Sr No 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 ∆ = 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 ∏t + 8 sin 300∏t –cos 100∏t Fn=300 Hz

2) x(t)= 2 cos 2000∏t+ 3 sin 6000∏t + 8 cos 12000∏t Fn=12KHz

3) x(t)= 4 cos 100∏t 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 2∏(10)t

X2(t)= cos 2∏(50)t

X3(t)= cos 2∏(90)t

X4(t)= cos 2∏(130)t

Q) Signal x1(t)=10cos2∏(1000)t+ 5 cos2∏(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 600∏t+ 2cos800∏t. 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 has one more type of classification.

a) Recursive Systems

b) Non-Recursive Systems
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 of the 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

6) Convolution Theorem

The Circular property states that if z

Convolution of two sequences in time domain corresponds to multiplication of its Z transform sequence in
frequency domain.

7) Correlation Property

The Correlation of two sequences states that if z

8) Initial value Theorem

Initial value theorem states that if z

9) Final value Theorem

Final value theorem states that if z

RELATIONSHIP BETWEEN FOURIER TRANSFORM AND Z TRANSFORM.

There is a close relationship between Z transform and Fourier transform. If we replace the complex variable
z by e –jω, then z transform is reduced to Fourier transform.

Z transform of sequence x(n) is given by

Fourier transform of sequence x(n) is given by

Complex variable z is expressed in polar form as Z= rejω where r= |z| and ω is ∟z. Thus we can be written
as
Thus, X(z) can be interpreted as Fourier Transform of signal sequence (x(n) r–n). Here r–n grows with n if
r<1 and decays with n if r>1. X(z) converges for |r|= 1. hence Fourier transform may be viewed as Z
transform of the sequence evaluated on unit circle. Thus The relationship between DFT and Z transform is
given by

The frequency ω=0 is along the positive Re(z) axis and the frequency ∏/2 is along the positive Im(z) axis.
Frequency ∏ is along the negative Re(z) axis and 3∏/2 is along the negative Im(z) axis.

INVERSE Z TRANSFORM (IZT)

The signal can be converted from time domain into z domain with the help of z transform (ZT). Similar
way the signal can be converted from z domain to time domain with the help of inverse z transform(IZT).
The inverse z transform can be obtained by using two different methods.

1) Partial fraction expansion Method (PFE) / Application of residue theorem

2) Power series expansion Method (PSE)

1. PARTIAL FRACTION EXPANSION METHOD

In this method X(z) is first expanded into sum of simple partial fraction.

The above equation can be written in partial fraction expansion form and find the coefficient AK and take
IZT.
2. RESIDUE THEOREM METHOD

In this method, first find G(z)= zn-1 X(Z) and find the residue of G(z) at various poles of X(z).

3. POWER-SERIES EXPANSION METHOD

The z transform of a discrete time signal x(n) is given as

Expanding the above terms we have

x(z) = …..+x(-2)Z2+ x(-1)Z+ x(0)+ x(1) Z-1 + x(2) Z2 +….. (2)

This is the expansion of z transform in power series form. Thus sequence x(n) is given as

x(n) ={ ….. ,x(-2),x(-1),x(0),x(1),x(2),…………..}.

Power series can be obtained directly or by long division method.

SOLVE USING “POWER SERIES EXPANSION“ METHOD

RECURSIVE ALGORITHM

Thus

X(0) = a0/b0
X(1) = 1/b0 [ a1- x(0) b1]

X(2) = 1/b0 [ a1- x(1) b1 - x(0) b2] ……………

SOLVE USING “RECURSIVE ALGORITHM“ METHOD

Example 2:Find the magnitude and phase plot of

Example 5:Find the inverse Z Transform
POLE –ZERO PLOT

X(z) is a rational function, that is a ratio of two polynomials in z-1 or z.

The roots of the denominator or the value of z for which X(z) becomes infinite, defines locations of the
poles. The roots of the numerator or the value of z forwhich X(z) becomes zero, defines locations of the
zeros.

ROC dos not contain any poles of X(z). This is because x(z) becomes infinite at the locations of
the poles. Only poles affect the causality and stability of the system.

CASUALTY CRITERIA FOR LSI SYSTEM

LSI system is causal if and only if the ROC the system function is exterior to

the circle. i. e |z| > r. This is the condition for causality of the LSI system in terms of z transform. (The
condition for LSI system to be causal is h(n) = 0 ….. n<0 )

STABILITY CRITERIA FOR LSI SYSTEM

Bounded input x(n) produces bounded output y(n) in the LSI system only if

With this condition satisfied, the system will be stable. The above equation states that the LSI
system is stable if its unit sample response is absolutely summable. This is necessary and sufficient
condition for the stability of LSI system.

Magnitudes of overall sum is less than the sum of magnitudes of individual sums.
If H(z) is evaluated on the unit circle | z-n|=|z|=1.

Hence LSI system is stable if and only if the ROC the system function includes the unit circle. i.e r < 1.
This is the condition for stability of the LSI system in terms of

z transform. Thus

For stable system |z| < 1

For unstable system |z| > 1

Marginally stable system |z| = 1

Poles inside unit circle gives stable system. Poles outside unit circle gives unstable system. Poles on unit
circle give marginally stable system.

A causal and stable system must have a system function that converges for |z| > r < 1.

STANDARD INVERSE Z TRANSFORMS

ONE SIDED Z TRANSFORM

Properties of one sided z transform are same as that of two sided z transform except shifting property.

1) Time delay

2) Time advance
Examples:

Q) Determine one sided z transform for following signals

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

SOLUTION OF DIFFERENTIAL EQUATION

One sided Z transform is very efficient tool for the solution of difference equations with nonzero initial
condition. System function of LSI system can be obtained from its difference equation

Z{ x(n-1) } = z-1 X(z) + x(-1)

Z{ x(n-2) } = z-2 X(z) + z-1 x(-1) + x(-2)

Similarly

Z{ x(n+1) } = z X(z) - z x(0)

Z{ x(n+2) } = z2 X(z) - z1 x(0) + x(1)

Difference equations are used to find out the relation between input and output sequences. It is also used
to relate system function H(z) and Z transform.

The transfer function H(ω) can be obtained from system function H(z) by putting z=ejω. Magnitude and
phase response plot can be obtained by putting various values of ω.
First order Difference Equation

y(n) = x(n) + a y(n-1)

where y(n) = Output Response of the recursive system x(n) =

Input signal

a= Scaling factor

y(n-1) = Unit delay to output. Now

we will start at n=0

n=0 y(0) = x(0) + a y(-1) ….(1)

n=1 y(1) = x(1) + a y(0) ….(2)

= x(1) + a [ x(0) + a y(-1) ]

hence = a2 y(-1) + a x(0) + x(1) ….(3)

The first part (A) is response depending upon initial condition.

The second Part (B) is the response of the system to an input signal.

Zero state response (Forced response) : Consider initial condition are zero. (System is relaxed at time
n=0) i.e y(-1) =0

Zero Input response (Natural response) : No input is forced as system is in non- relaxed initial
condition. i.e y(-1) != 0

Total response is the sum of zero state response and zero input response.

Q) Determine zero input response for y(n) – 3y(n-1) – 4y(n-2)=0; (Initial Conditions are y(-1)=5 & y(-2)=
10) Answer: y(n)= 7 (-1)n + 48 (4)n

Q) A difference equation of the system is given below

Y(n)= 0.5 y(n-1) + x(n)

Determine a) System function

 b) Pole zero plot

Unit sample response

A difference equation of the system is given below Y(n)= 0.7 y(n-1) – 0.12 y(n-2) + x(n-1) + x(n-2)

System Function b) Pole zero plot

Response of system to the input x(n) = nu(n)

Is the system stable? Comment on the result.

Q) A difference equation of the system is given below

Y(n)= 0.5 x(n) + 0.5 x(n-1)

Determine

a) System function

b) Pole zero plot

Unit sample response

Transfer function

Magnitude and phase plot

A difference equation of the system is given below a. Y(n)= 0.5 y(n-1) + x(n) + x(n-1)

Y(n)= x(n) + 3x(n-1) + 3x(n-2) + x(n-3)

System Function b) Pole zero plot

Unit sample response

Find values of y(n) for n=0,1,2,3,4,5 for x(n)= δ(n) for no initial condition.

Q) Solve second order difference equation

2x(n-2) – 3x(n-1) + x(n) = 3n-2 with x(-2)=-4/9 and x(-1)=-1/3.

Q) Solve second order difference equation x(n+2) + 3x(n+1) + 2x(n) with x(0)=0 and x(1)=1.

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

6) Convolution Theorem

The Circular property states that if z

Convolution of two sequences in time domain corresponds to multiplication of its Z transform sequence in
frequency domain.

7) Correlation Property

The Correlation of two sequences states that if z

8) Initial value Theorem

Initial value theorem states that if z

9) Final value Theorem

Final value theorem states that if z

RELATIONSHIP BETWEEN FOURIER TRANSFORM AND Z TRANSFORM.

There is a close relationship between Z transform and Fourier transform. If we replace the complex variable
z by e –jω, then z transform is reduced to Fourier transform.

Z transform of sequence x(n) is given by

Fourier transform of sequence x(n) is given by

Complex variable z is expressed in polar form as Z= rejω where r= |z| and ω is ∟z. Thus we can be written
as
Thus, X(z) can be interpreted as Fourier Transform of signal sequence (x(n) r–n). Here r–n grows with n if
r<1 and decays with n if r>1. X(z) converges for |r|= 1. hence Fourier transform may be viewed as Z
transform of the sequence evaluated on unit circle. Thus The relationship between DFT and Z transform is
given by

The frequency ω=0 is along the positive Re(z) axis and the frequency ∏/2 is along the positive Im(z) axis.
Frequency ∏ is along the negative Re(z) axis and 3∏/2 is along the negative Im(z) axis.

INVERSE Z TRANSFORM (IZT)

The signal can be converted from time domain into z domain with the help of z transform (ZT). Similar
way the signal can be converted from z domain to time domain with the help of inverse z transform(IZT).
The inverse z transform can be obtained by using two different methods.

1) Partial fraction expansion Method (PFE) / Application of residue theorem

2) Power series expansion Method (PSE)

1. PARTIAL FRACTION EXPANSION METHOD

In this method X(z) is first expanded into sum of simple partial fraction.

The above equation can be written in partial fraction expansion form and find the coefficient AK and take
IZT.
2. RESIDUE THEOREM METHOD

In this method, first find G(z)= zn-1 X(Z) and find the residue of G(z) at various poles of X(z).

3. POWER-SERIES EXPANSION METHOD

The z transform of a discrete time signal x(n) is given as

Expanding the above terms we have

x(z) = …..+x(-2)Z2+ x(-1)Z+ x(0)+ x(1) Z-1 + x(2) Z2 +….. (2)

This is the expansion of z transform in power series form. Thus sequence x(n) is given as

x(n) ={ ….. ,x(-2),x(-1),x(0),x(1),x(2),…………..}.

Power series can be obtained directly or by long division method.

LINEAR CONVOLUTION SUM METHOD

1. This method is powerful analysis tool for studying LSI Systems.

2. In this method we decompose input signal into sum of elementary signal. Now the elementary input
signals are taken into account and individually given to the system. Now using linearity property whatever
output response we get for decomposed input signal, we simply add it & this will provide us total response
of the system to any given input signal.

3. Convolution involves folding, shifting, multiplication and summation operations.

4. 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 equals to M+n-1.

Linear Convolution states that

y(n) = x(n) * h(n)

Example 1: h(n) = { 1 , 2 , 1, -1 } & x(n) = { 1, 2, 3, 1 } Find y(n)

METHOD 1: GRAPHICAL REPRESENTATION

Step 1) Find the value of n = nx+ nh = -1 (Starting Index of x(n)+ starting index of h(n))

Step 2) y(n)= { y(-1) , y(0) , y(1), y(2), ….} It goes up to length(xn)+ length(yn) -1. i.e n=-1

METHOD 2: MATHEMATICAL FORMULA

Use Convolution formula

k= 0 to 3 (start index to end index of x(n))

y(n) = x(0) h(n) + x(1) h(n-1) + x(2) h(n-2) + x(3) h(n-3)

METHOD 3: VECTOR FORM (TABULATION METHOD)

X(n)= {x1,x2,x3} & h(n) ={ h1,h2,h3}

y(-1) = h1 x1

y(0) = h2 x1 + h1 x2

y(1) = h1 x3 + h2x2 + h3 x1 …………

METHOD 4: SIMPLE MULTIPLICATION FORM

PROPERTIES OF LINEAR CONVOLUTION

x(n) = Excitation Input signal y(n)

= Output Response

h(n) = Unit sample response

1. Commutative Law: (Commutative Property of Convolution)

x(n) * h(n) = h(n) * x(n)

2. Associate Law: (Associative Property of Convolution)

[ x(n) * h1(n) ] * h2(n) = x(n) * [ h1(n) * h2(n) ]

3. Distribute Law: (Distributive property of convolution)

x(n) * [ h1(n) + h2(n) ] = x(n) * h1(n) + x(n) * h2(n)

CAUSALITY OF LSI SYSTEM

The output of causal system depends upon the present and past inputs. The output of the causal system at
n= n0 depends only upon inputs x(n) for n≤ n0. The linear convolution is given as

The output of causal system at n= n0 depends upon the inputs for n< n0 Hence h(-1)=h(-2)=h(-3)=0

Thus LSI system is causal if and only if

h(n) =0 for n<0

This is the necessary and sufficient condition for causality of the system. Linear convolution of the causal
LSI system is given by
STABILITY FOR LSI SYSTEM

A System is said to be stable if every bounded input produces a bounded output.

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 < ∞.

Linear convolution is given by

Taking the absolute value of both sides

The absolute values of total sum is always less than or equal to sum of the absolute values of individually
terms. Hence

The input x(n) is said to bounded if there exists some finite number Mx such that |x(n)| ≤ Mx < ∞.

Hence bounded input x(n) produces bounded output y(n) in the LSI system only if

With this condition satisfied, the system will be stable. The above equation states that the LSI system is
stable if its unit sample response is absolutely summable. This is necessary and sufficient condition for the
stability of LSI system.

Example 1:
SELF-STUDY: Exercise No. 1

Q1) Show that the discrete time signal is periodic only if its frequency is expressed as the ratio of two
integers.

Q2) Show that the frequency range for discrete time sinusoidal signal is -∏ to ∏ radians/sample or -½
cycles/sample to ½ cycles/sample.

Q3) Prove δ (n)= u(n)= u(n-1)

Q6) Prove that every discrete sinusoidal signal can be expressed in terms of weighted unit impulse. Q7)
Prove the Linear Convolution theorem.

CORRELATION:

It is frequently necessary to establish similarity between one set of data and another. It means we would
like to correlate two processes or data. Correlation is closely related to convolution, because the correlation
is essentially convolution of two data sequences in which one of the sequences has been reversed.

Applications are in

1. Images processing for robotic vision or remote sensing by satellite in which data from different image
is compared

2. In radar and sonar systems for range and position finding in which transmitted and reflected waveforms
are compared.

3. Correlation is also used in detection and identifying of signals in noise.

4. Computation of average power in waveforms.

5. Identification of binary codeword in pulse code modulation system.

DIFFERENCE BETWEEN LINEAR CONVOLUTION AND CORRELATION

Linear Convolution

1. In case of convolution two signal sequences input signal and impulse response given by the same system
is calculated

2. Our main aim is to calculate the response given by the system.

3. Linear Convolution is given by the equation y(n) = x(n) * h(n) & calculated as

4. Linear convolution is commutative

Correlation

1. In case of Correlation, two signal sequences are just compared.

2. Our main aim is to measure the degree to which two signals are similar and thus to extract some
information that depends to a large extent on the application

3. Received signal sequence is given as Y(n) = α x(n-D) + ω(n) Where α= Attenuation Factor D= Delay
ω(n) = Noise signal
4. Not commutative.

TYPES OF CORRELATION

Under Correlation there are two classes.

1) CROSS CORRELATION: When the correlation of two different sequences x(n) and y(n) is performed
it is called as Cross correlation. Cross-correlation of x(n) and y(n) is rxy(l) which can be mathematically
expressed as

2) AUTO CORRELATION: In Auto-correlation we correlate signal x(n) with itself, which can be
mathematically expressed as

PROPERTIES OF CORRELATION

1) The cross-correlation is not commutative.

rxy(l) = ryx(-l)
2) The cross-correlation is equivalent to convolution of one sequence with folded version of another
sequence.

rxy(l) = x(l) * y(-l).

3) The autocorrelation sequence is an even function. rxx(l) = rxx(-l)

Examples:

Q) Determine cross-correlation sequence

x(n)={2, -1, 3, 7,1,2, -3} & y(n)={1, -1, 2, -2, 4, 1, -2 ,5}

Answer: rxy(l) = {10, -9, 19, 36, -14, 33, 0,7, 13, -18, 16, -7, 5, -3}

Q) Determine autocorrelation sequence

x(n)={1, 2, 1, 1} Answer: rxx(l) = {1, 3, 5, 7, 5, 3, 1}

DIFFERENCE BETWEEN LINEAR CONVOLUTION AND CORRELATION

Linear Convolution
1. In case of convolution two signal sequences input signal and impulse response given by the same system
is calculated

2. Our main aim is to calculate the response given by the system.

3. Linear Convolution is given by the equation y(n) = x(n) * h(n) & calculated as

4. Linear convolution is commutative

Correlation

1. In case of Correlation, two signal sequences are just compared.

2. Our main aim is to measure the degree to which two signals are similar and thus to extract some
information that depends to a large extent on the application

3. Received signal sequence is given as Y(n) = α x(n-D) + ω(n) Where α= Attenuation Factor D= Delay
ω(n) = Noise signal

4. Not commutative.

FREQUENCY TRANSFORMATIONS

INTRODUCTION

Any signal can be decomposed in terms of sinusoidal (or complex exponential) components. Thus the
analysis of signals can be done by transforming time domain signals into frequency domain and vice-versa.
This transformation between time and frequency domain is performed with the help of Fourier
Transform(FT) But still it is not convenient for computation by DSP processors hence Discrete Fourier
Transform(DFT) is used.

Time domain analysis provides some information like amplitude at sampling instant but does not convey
frequency content & power, energy spectrum hence frequency domain analysis is used.

For Discrete time signals x(n) , Fourier Transform is denoted as x(ω) & given by
DIFFERENCE BETWEEN FT & DFT

CALCULATION OF DFT & IDFT

For calculation of DFT & IDFT two different methods can be used. First method is using mathematical
equation & second method is 4 or 8 point DFT. If x(n) is the sequence of N samples then consider WN= e –
j2 ∏ / N
(twiddle factor)

Four POINT DFT ( 4-DFT)

EIGHT POINT DFT ( 8-DFT)

Examples:

Q) Compute DFT of x(n) = {0,1,2,3} Ans: x4=[6, -2+2j, -2, -2-2j ]

Q) Compute DFT of x(n) = {1,0,0,1} Ans: x4=[2, 1+j, 0, 1-j ]

Q) Compute DFT of x(n) = {1,0,1,0} Ans: x4=[2, 0, 2, 0 ]

Q) Compute IDFT of x(k) = {2, 1+j, 0, 1-j } Ans: x4=[1,0,0,1]

DIFFERENCE BETWEEN DFT & IDFT

PROPERTIES OF DFT

1. Periodicity

Let x(n) and x(k) be the DFT pair then if

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

X(k+N) = X(k) for all k

Thus periodic sequence xp(n) can be given as

2. Linearity

The linearity property states that if

DFT of linear combination of two or more signals is equal to the same linear combination of DFT of
individual signals.
3. Circular Symmetries of a sequence

A) A sequence is said to be circularly even if it is symmetric about the point zero on the circle. Thus X(N-
n) = x(n)

B) A sequence is said to be circularly odd if it is anti symmetric about the point zero on the circle. Thus
X(N-n) = - x(n)

C) A circularly folded sequence is represented as x((-n))N and given by x((-n))N = x(N-n).

D) Anticlockwise direction gives delayed sequence and clockwise direction gives advance sequence.

Thus delayed or advances sequence x`(n) is related to x(n) by the circular shift.

4. Symmetry Property of a sequence

A. Symmetry property for real valued x(n) i.e xI(n)=0

This property states that if x(n) is real then X(N-k) = X*(k)=X(-k)

B) Real and even sequence x(n) i.e xI(n)=0 & XI(K)=0

This property states that if the sequence is real and even x(n)= x(N-n) then DFT becomes N-1

C) Real and odd sequence x(n) i.e xI(n)=0 & XR(K)=0

This property states that if the sequence is real and odd x(n)=-x(N-n) then DFT becomes N-1

D) Pure Imaginary x(n) i.e xR(n)=0

This property states that if the sequence is purely imaginary x(n)=j XI(n) then DFT becomes

5. Circular Convolution

The Circular Convolution property states that if

It means that circular convolution of x1(n) & x2(n) is equal to multiplication of their DFT s. Thus circular
convolution of two periodic discrete signal with period N is given by

Multiplication of two sequences in time domain is called as Linear convolution while Multiplication of two
sequences in frequency domain is called as circular convolution. Results of both are totally different but
are related with each other.

There are two different methods are used to calculate circular convolution

Graphical representation form

Matrix approach

DIFFERENCE BETWEEN LINEAR CONVOLUTION & CIRCULAR CONVOLUTION

Linear Convolution

1. In case of convolution two signal sequences input signal x(n) and impulse response h(n) given by the
same system, output y(n) is calculated

2. Multiplication of two sequences in time domain is called as Linear convolution

3. Linear Convolution is given by the equation y(n) = x(n) * h(n) & calculated as

4. Linear Convolution of two signals returns N-1 elements where N is sum of elements in both sequences.

Circular Convolution

1. Multiplication of two DFT s is called as circular convolution.

2. Multiplication of two sequences in frequency domain is called as circular convolution.

3. Circular Convolution is calculated as

4. Circular convolution returns same number of elements that of two signals.

Q) The two sequences x1(n)={2,1,2,1} & x2(n)={1,2,3,4}. Find out the sequence x3(m) which is equal to
circular convolution of two sequences. Ans: X3(m)={14,16,14,16}

Q) x1(n)={1,1,1,1,-1,-1,- 1,-1} & x2(n)={0,1,2,3,4,3,2,1}. Find out the sequence x3(m) which is equal to
circular convolution of two sequences. Ans: X3(m)={-4,-8,-8,-4,4,8,8,4}

Q) Perform Linear Convolution of x(n)={1,2} & h(n)={2,1} using DFT & IDFT.

Q) Perform Linear Convolution of x(n)={1,2,2,1} & h(n)={1,2,3} using 8 Pt DFT & IDFT.

DIFFERENCE BETWEEN LINEAR CONVOLUTION & CIRCULAR CONVOLUTION

6. Multiplication

The Multiplication property states that if

It means that multiplication of two sequences in time domain results in circular convolution of their DFT s
in frequency domain.

7. Time reversal of a sequence

The Time reversal property states that if

It means that the sequence is circularly folded its DFT is also circularly folded.

8. Circular Time shift

The Circular Time shift states that if

Thus shifting the sequence circularly by „l samples is equivalent to multiplying its DFT by e –j2 ∏ k l / N
9. Circular frequency shift

The Circular frequency shift states that if

Thus shifting the frequency components of DFT circularly is equivalent to multiplying its time domain
sequence by e –j2 ∏ k l / N

10. Complex conjugate property

The Complex conjugate property states that if

11. Circular Correlation

The Complex correlation property states

Here rxy(l) is circular cross correlation which is given as

This means multiplication of DFT of one sequence and conjugate DFT of another sequence is equivalent
to circular cross-correlation of these sequences in time domain.

12.Parseval’sTheorem

The Parseval s theorem states

This equation give energy of finite duration sequence in terms of its frequency components.

APPLICATION OF DFT

1. DFT FOR LINEAR FILTERING

Consider that input sequence x(n) of Length L & impulse response of same system is h(n) having M
samples. Thus y(n) output of the system contains N samples where N=L+M-1. If DFT of y(n) also contains
N samples then only it uniquely represents y(n) in time domain. Multiplication of two DFT s is equivalent
to circular convolution of corresponding time domain sequences. But the length of x(n) & h(n) is less than
N. Hence these sequences are appended with zeros to make their length N called as “Zero padding”. The N
point circular convolution and linear convolution provide the same sequence. Thus linear convolution can
be obtained by circular convolution. Thus linear filtering is provided by DFT.

When the input data sequence is long then it requires large time to get the output sequence. Hence other
techniques are used to filter long data sequences. Instead of finding the output of complete input sequence
it is broken into small length sequences. The output due to these small length sequences are computed fast.
The outputs due to these small length sequences are fitted one after another to get the final output response.
METHOD 1: OVERLAP SAVE METHOD OF LINEAR FILTERING

Step 1> In this method L samples of the current segment and M-1 samples of the previous segment forms
the input data block. Thus data block will be

X1(n) ={0,0,0,0,0,………………… ,x(0),x(1),…………….x(L-1)}

X2(n) ={x(L-M+1), …………….x(L-1),x(L),x(L+1),,,,,,,,,,,,,x(2L-1)} X3(n) ={x(2L-M+1),

…………….x(2L-1),x(2L),x(2L+2),,,,,,,,,,,,,x(3L-1)}

Step2> Unit sample response h(n) contains M samples hence its length is made N by padding zeros. Thus
h(n) also contains N samples.

h(n)={ h(0), h(1), …………….h(M-1), 0,0,0,……………………(L-1 zeros)}

Step3> The N point DFT of h(n) is H(k) & DFT of mth data block be xm(K) then corresponding DFT of
output be Y`m(k)

Y`m(k)= H(k) xm(K)

Step 4> The sequence ym(n) can be obtained by taking N point IDFT of Y`m(k). Initial (M-1) samples in
the corresponding data block must be discarded. The last L samples are the correct output samples. Such
blocks are fitted one after another to get the final output.
METHOD 2: OVERLAP ADD METHOD OF LINEAR FILTERING

Step 1> In this method L samples of the current segment and M-1 samples of the previous segment forms
the input data block. Thus data block will be

X1(n) ={x(0),x(1),…………….x(L-1),0,0,0,……….}
X2(n) ={x(L),x(L+1),x(2L-1),0,0,0,0}

X3(n) ={x(2L),x(2L+2),,,,,,,,,,,,,x(3L-1),0,0,0,0}

Step2> Unit sample response h(n) contains M samples hence its length is made N by padding zeros. Thus
h(n) also contains N samples.

h(n)={ h(0), h(1), …………….h(M-1), 0,0,0,……………………(L-1 zeros)}

Step3> The N point DFT of h(n) is H(k) & DFT of mth data block be xm(K) then corresponding DFT of
output be Y`m(k)

Y`m(k)= H(k) xm(K)

Step 4> The sequence ym(n) can be obtained by taking N point IDFT of Y`m(k). Initial (M-1) samples are
not discarded as there will be no aliasing. The last (M-1) samples of current output block must be added to
the first M-1 samples of next output block. Such blocks are fitted one after another to get the final output.
DIFFERENCE BETWEEN OVERLAP SAVE AND OVERLAP ADD METHOD
OVERLAP SAVE METHOD

1. In this method, L samples of the current segment and (M-1) samples of the previous segment forms
the input data block.

2. Initial M-1 samples of output sequence are discarded which occurs due to aliasing effect.

3. To avoid loss of data due to aliasing last M-1 samples of each data record are saved.

OVERLAP ADD METHOD

1. In this method L samples from input sequence and padding M-1 zeros forms data block of size N.

2. There will be no aliasing in output data blocks.

3. Last M-1 samples of current output block must be added to the first M-1 samples of next output
block. Hence called as overlap add method.

2. SPECTRUM ANALYSIS USING DFT

DFT of the signal is used for spectrum analysis. DFT can be computed on digital computer or digital signal
processor. The signal to be analyzed is passed through anti-aliasing filter and samples at the rate of Fs≥ 2
Fmax. Hence highest frequency component is Fs/2.

Frequency spectrum can be plotted by taking N number of samples & L samples of waveforms. The total
frequency range 2∏ is divided into N points. Spectrum is better if we take large value of N & L But this
increases processing time. DFT can be computed quickly using FFT algorithm hence fast processing can
be done. Thus most accurate resolution can be obtained by increasing number of samples.

FAST FOURIER ALGORITHM (FFT)

1. Large number of the applications such as filtering, correlation analysis, spectrum analysis require
calculation of DFT. But direct computation of DFT require large number of computations and hence
processor remain busy. Hence special algorithms are developed to compute DFT quickly called as Fast
Fourier algorithms (FFT).

2. The radix-2 FFT algorithms are based on divide and conquer approach. In this method, the N-point DFT
is successively decomposed into smaller DFT s. Because of this decomposition, the number of
computations are reduced.

RADIX-2 FFT ALGORITHMS

DECIMATION IN TIME (DITFFT)

There are three properties of twiddle factor WN

N point sequence x(n) be splitted into two N/2 point data sequences f1(n) and f2(n). f1(n) contains even
numbered samples of x(n) and f2(n) contains odd numbered samples of x(n). This splitted operation is
called decimation. Since it is done on time domain sequence it is called “Decimation in Time”. Thus

f1(m)=x(2m) where n=0,1,………….N/2-1

f2(m)=x(2m+1) where n=0,1,………….N/2-1

N point DFT is given as

Since the sequence x(n) is splitted into even numbered and odd numbered samples, thus

Fig 1 shows that 8-point DFT can be computed directly and hence no reduction in computation.
COMPUTATIONAL COMPLEXITY FFT V/S DIRECT COMPUTATION

For Radix-2 algorithm value of N is given as N= 2V

Hence value of v is calculated as

V= log10 N / log10 2 = log2 N

Thus if value of N is 8 then the value of v=3. Thus three stages of decimation. Total number of butterflies
will be Nv/2 = 12.

If value of N is 16 then the value of v=4. Thus four stages of decimation. Total number of butterflies will
be Nv/2 = 32.
Each butterfly operation takes two addition and one multiplication operations. Direct computation requires
N2 multiplication operation & N2 – N addition operations.

MEMORY REQUIREMENTS AND IN PLACE COMPUTATION

From values a and b new values A and B are computed. Once A and B are computed, there is no need to
store a and b. Thus same memory locations can be used to store A and B where a and b were stored hence
called as In place computation. The advantage of in place computation is that it reduces memory
requirement.

Thus for computation of one butterfly, four memory locations are required for storing two complex numbers
A and B. In every stage there are N/2 butterflies hence total 2N memory locations are required. 2N locations
are required for each stage. Since stages are computed successively these memory locations can be shared.
In every stage N/2 twiddle factors are required hence maximum storage requirements of N point DFT will
be (2N + N/2).

BIT REVERSAL
For 8 point DIT DFT input data sequence is written as x(0), x(4), x(2), x(6), x(1), x(5), x(3), x(7) and the
DFT sequence X(k) is in proper order as X(0), X(1), X(2), X(3), X(4), x(5), X(6), x(7). In DIF FFT it is
exactly opposite. This can be obtained by bit reversal method.

Table shows first column of memory address in decimal and second column as binary. Third column
indicates bit reverse values. As FFT is to be implemented on digital computer simple integer division by 2
method is used for implementing bit reversal algorithms. Flow chart for Bit reversal algorithm is as follows
DECIMATION IN FREQUENCY (DIFFFT)

In DIF N Point DFT is splitted into N/2 points DFT s. X(k) is splitted with k even and k odd this is called
Decimation in frequency(DIF FFT).

N point DFT is given as

Since the sequence x(n) is splitted N/2 point samples, thus

Let us split X(k) into even and odd numbered samples
Fig 2 shows signal flow graph and stages for computation of radix-2 DIF FFT algorithm of N=4

Fig 3 shows signal flow graph and stages for computation of radix-2 DIF FFT algorithm of N=8
DIFFERENCE BETWEEN DITFFT AND DIFFFT

DIT FFT

1. DITFFT algorithms are based upon decomposition of the input sequence into smaller and smaller sub
sequences.

2. In this input sequence x(n) is splitted into even and odd numbered samples

3. Splitting operation is done on time domain sequence.

4. In DIT FFT input sequence is in bit reversed order while the output sequence is in natural order.

DIF FFT

1. DIFFFT algorithms are based upon decomposition of the output sequence into smaller and smaller sub
sequences.

2. In this output sequence X(k) is considered to be splitted into even and odd numbered samples

3. Splitting operation is done on frequency domain sequence.

4. In DIFFFT, input sequence is in natural order. And DFT should be read in bit reversed order.
DIFFERENCE BETWEEN DIRECT COMPUTATION & FFT

Direct Computation

1. Direct computation requires large number of computations as compared with FFT algorithms.

2. Processing time is more and more for large number of N hence processor remains busy.

3. Direct computation does not requires splitting operation.

4. As the value of N in DFT increases, the efficiency of direct computation decreases.

5. In those applications where DFT is to be computed only at selected values of k(frequencies) and when
these values are less than log2N then direct computation becomes more efficient than FFT.

Radix -2 FFT Algorithms

1. Radix-2 FFT algorithms requires less number of computations.

2. Processing time is less hence these algorithms compute DFT very quickly as compared with direct
computation.

3. Splitting operation is done on time domain basis (DIT) or frequency domain basis (DIF)
4. As the value of N in DFT increases, the efficiency of FFT algorithms increases.

5. Applications 1) Linear filtering 2) Digital filter design

Q) x(n)={1,2,2,1} Find X(k) using DITFFT.

Q) x(n)={1,2,2,1} Find X(k)using DIFFFT.

Q) x(n)={0.3535,0.3535,0.6464,1.0607,0.3535,-1.0607,-1.3535,-0.3535} Find X(k) using DITFFT.

Q) Using radix 2 FFT algorithm, plot flow graph for N=8.

GOERTZEL ALGORITHM

FFT algorithms are used to compute N point DFT for N samples of the sequence x(n). This requires N/2
log2N number of complex multiplications and N log2N complex additions. In some applications DFT is to
be computed only at selected values of frequencies and selected values are less than log2N, then direct
computations of DFT becomes more efficient than FFT. This direct computations of DFT can be realized
through linear filtering of x(n). Such linear filtering for computation of DFT can be implemented using
Goertzel algorithm.

By definition N point DFT is given as

Thus DFT can be obtained as the output of LSI system at n=N. Such systems can give X(k) at selected
values of k. Thus DFT is computed as linear filtering operations by Goertzel Algorithm.

IIR FILTER DESIGN

INTRODUCTION

To remove or to reduce strength of unwanted signal like noise and to improve the quality of required signal
filtering process is used. To use the channel full bandwidth we mix up two or more signals on transmission
side and on receiver side we would like to separate it out in efficient way.

Hence filters are used. Thus the digital filters are mostly used in
1. Removal of undesirable noise from the desired signals

2. Equalization of communication channels

3. Signal detection in radar, sonar and communication

4. Performing spectral analysis of signals.

Analog and digital filters

In signal processing, the function of a filter is to remove unwanted parts of the signal, such as random noise,
or to extract useful parts of the signal, such as the components lying within a certain frequency range.

The following block diagram illustrates the basic idea.

There are two main kinds of filter, analog and digital. They are quite different in their physical makeup and
in how they work.
An analog filter uses analog electronic circuits made up from components such as resistors, capacitors and
op amps to produce the required filtering effect. Such filter circuits are widely used in such applications as
noise reduction, video signal enhancement, graphic equalizers in hi-fi systems, and many other areas.

In analog filters the signal being filtered is an electrical voltage or current which is the direct analogue of
the physical quantity (e.g. a sound or video signal or transducer output) involved.

A digital filter uses a digital processor to perform numerical calculations on sampled values of the signal.
The processor may be a general-purpose computer such as a PC, or a specialized DSP (Digital Signal
Processor) chip.

The analog input signal must first be sampled and digitized using an ADC (analog to digital converter). The
resulting binary numbers, representing successive sampled values of the input signal, are transferred to the
processor, which carries out numerical calculations on them. These calculations typically involve
multiplying the input values by constants and adding the products together. If necessary, the results of these
calculations, which now represent sampled values of the filtered signal, are output through a DAC (digital
to analog converter) to convert the signal back to analog form.

In a digital filter, the signal is represented by a sequence of numbers, rather than a voltage or current.

The following diagram shows the basic setup of such a system.

BASIC BLOCK DIAGRAM OF DIGITAL FILTERS

a. Samplers are used for converting continuous time signal into a discrete time signal by taking samples
of the continuous time signal at discrete time instants.

b. The Quantizer are used for converting a discrete time continuous amplitude signal into a digital signal
by expressing each sample value as a finite number of digits.

c. In the encoding operation, the quantization sample value is converted to the binary equivalent of that
quantization level.

d. The digital filters are the discrete time systems used for filtering of sequences.

e. These digital filters performs the frequency related operations such as low pass, high pass, band pass
and band reject etc. These digital Filters are designed with digital hardware and software and are represented
by difference equation.

DIFFERENCE BETWEEN ANALOG FILTER AND DIGITAL FILTER

Analog Filter

1 Analog filters are used for filtering analog signals.

2 Analog filters are designed with various components like resistor, inductor and capacitor

3 Analog filters less accurate & because of component tolerance of active components & more sensitive
to environmental changes.

4 Less flexible

5 Filter representation is in terms of system components.

6 An analog filter can only be changed by redesigning the filter circuit.

Digital Filter

1 Digital filters are used for filtering digital sequences.

2 Digital Filters are designed with digital hardware like FF, counters shift registers, ALU and software s
like C or assembly language.
3 Digital filters are less sensitive to the environmental changes, noise and disturbances. Thus periodic
calibration can be avoided. Also they are extremely stable.

4 These are most flexible as software programs & control programs can be easily modified. Several input
signals can be filtered by one digital filter.

5 Digital filters are represented by the difference equation.

6 A digital filter is programmable, i.e. its operation is determined by a program stored in the processor's
memory. This means the digital filter can easily be changed without affecting the circuitry (hardware).

FILTER TYPES AND IDEAL FILTER CHARACTERISTIC

Filters are usually classified according to their frequency-domain characteristic as lowpass, highpass,
bandpass and bandstop filters.

Lowpass Filter

A lowpass filter is made up of a passband and a stopband, where the lower frequencies Of the input signal
are passed through while the higher frequencies are attenuated.

Highpass Filter

A highpass filter is made up of a stopband and a passband where the lower frequencies of the input signal
are attenuated while the higher frequencies are passed.
Bandpass Filter

A bandpass filter is made up of two stopbands and one passband so that the lower and higher frequencies
of the input signal are attenuated while the intervening frequencies are passed.

Bandstop Filter

A bandstop filter is made up of two passbands and one stopband so that the lower and higher frequencies
of the input signal are passed while the intervening frequencies are attenuated. An idealized bandstop filter
frequency response has the following
Multipass Filter

A multipass filter begins with a stopband followed by more than one passband. By default, a multipass filter
in Digital Filter Designer consists of three passbands and

four stopbands. The frequencies of the input signal at the stopbands are attenuated while those at the
passbands are passed.

Multistop Filter

A multistop filter begins with a passband followed by more than one stopband. By default, a multistop filter
in Digital Filter Designer consists of three passbands and two stopbands.

All Pass Filter

An all pass filter is defined as a system that has a constant magnitude response for all frequencies.

|H(ω)| = 1 for 0 ≤ ω < ∏

The simplest example of an all pass filter is a pure delay system with system function H(z) = Z-k. This is a
low pass filter that has a linear phase characteristic.

All Pass filters find application as phase equalizers. When placed in cascade with a system that has an
undesired phase response, a phase equalizers is designed to
compensate for the poor phase characteristic of the system and therefore to produce an overall linear phase
response.

IDEAL FILTER CHARACTERISTIC

1. Ideal filters have a constant gain (usually taken as unity gain) passband characteristic and zero gain in
their stop band.

2. Ideal filters have a linear phase characteristic within their passband.

3. Ideal filters also have constant magnitude characteristic.

4. Ideal filters are physically unrealizable.

DIFFERENCE BETWEEN ANALOG FILTER AND DIGITAL FILTER

Analog Filter
1 Analog filters are used for filtering analog signals.

2 Analog filters are designed with various components like resistor, inductor and capacitor

3 Analog filters less accurate & because of component tolerance of active components & more sensitive
to environmental changes.

4 Less flexible

5 Filter representation is in terms of system components.

6 An analog filter can only be changed by redesigning the filter circuit.

Digital Filter

1 Digital filters are used for filtering digital sequences.

2 Digital Filters are designed with digital hardware like FF, counters shift registers, ALU and software s
like C or assembly language.

3 Digital filters are less sensitive to the environmental changes, noise and disturbances. Thus periodic
calibration can be avoided. Also they are extremely stable.

4 These are most flexible as software programs & control programs can be easily modified. Several input
signals can be filtered by one digital filter.

5 Digital filters are represented by the difference equation.

6 A digital filter is programmable, i.e. its operation is determined by a program stored in the processor's
memory. This means the digital filter can easily be changed without affecting the circuitry (hardware).

FILTER TYPES AND IDEAL FILTER CHARACTERISTIC

Filters are usually classified according to their frequency-domain characteristic as lowpass, highpass,
bandpass and bandstop filters.

1. Lowpass Filter

A lowpass filter is made up of a passband and a stopband, where the lower frequencies Of the input signal
are passed through while the higher frequencies are attenuated.
2. Highpass Filter

A highpass filter is made up of a stopband and a passband where the lower frequencies of the input signal
are attenuated while the higher frequencies are passed.

3. Bandpass Filter

A bandpass filter is made up of two stopbands and one passband so that the lower and higher frequencies
of the input signal are attenuated while the intervening frequencies are passed.
4. Bandstop Filter

A bandstop filter is made up of two passbands and one stopband so that the lower and higher frequencies
of the input signal are passed while the intervening frequencies are attenuated. An idealized bandstop filter
frequency response has the following

5. Multipass Filter

A multipass filter begins with a stopband followed by more than one passband. By default, a multipass filter
in Digital Filter Designer consists of three passbands and

four stopbands. The frequencies of the input signal at the stopbands are attenuated while those at the
passbands are passed.

6. Multistop Filter
A multistop filter begins with a passband followed by more than one stopband. By default, a multistop filter
in Digital Filter Designer consists of three passbands and two stopbands.

7. All Pass Filter

An all pass filter is defined as a system that has a constant magnitude response for all frequencies.

|H(ω)| = 1 for 0 ≤ ω < ∏

The simplest example of an all pass filter is a pure delay system with system function H(z) = Z-k. This is a
low pass filter that has a linear phase characteristic.

All Pass filters find application as phase equalizers. When placed in cascade with a system that has an
undesired phase response, a phase equalizers is designed to

compensate for the poor phase characteristic of the system and therefore to produce an overall linear phase
response.

IDEAL FILTER CHARACTERISTIC

1. Ideal filters have a constant gain (usually taken as unity gain) passband characteristic and zero gain in
their stop band.

2. Ideal filters have a linear phase characteristic within their passband.

3. Ideal filters also have constant magnitude characteristic.

4. Ideal filters are physically unrealizable.

TYPES OF DIGITAL FILTER

Digital filters are of two types. Finite Impulse Response Digital Filter & Infinite Impulse Response Digital
Filter

DIFFERENCE BETWEEN FIR FILTER AND IIR FILTER

FIR Digital Filter

1. FIR system has finite duration unit sample response. i.e h(n) = 0 for n<0 and n ≥ M Thus the unit sample
response exists for the duration from 0 to M-1.
2. FIR systems are non recursive. Thus output of FIR filter depends upon present and past inputs.

3. Difference equation of the LSI system for FIR filters becomes

4. FIR systems has limited or finite memory requirements.

5. FIR filters are always stable

6. FIR filters can have an exactly linear phase response so that no phase distortion is introduced in the signal
by the filter.

7. The effect of using finite word length to implement filter, noise and quantization errors are less severe in
FIR than in IIR.

8. All zero filters

9. FIR filters are generally used if no phasedistortion is desired.

Example:

System described by

Y(n) = 0.5 x(n) + 0.5 x(n-1) is FIR filter.

h(n)={0.5,0.5}

IIR Digital Filter

1. IIR system has infinite duration unit sample response. i. e h(n) = 0 for n<0 Thus the unit sample response
exists for the duration from 0 to ∞.

2. IIR systems are recursive. Thus they use feedback. Thus output of IIR filter depends upon present and
past inputs as well as past outputs

3. Difference equation of the LSI system for IIR filters becomes

4. IIR system requires infinite memory.

5. Stability cannot be always guaranteed.

6. IIR filter is usually more efficient design in terms of computation time and memory requirements. IIR
systems usually requires less processing time and storage as compared with FIR.

7. Analogue filters can be easily and readily transformed into equivalent IIR digital filter. But same is not
possible in FIR because that have no analogue counterpart.

8. Poles as well as zeros are present.

9. IIR filters are generally used if sharp cutoff and high throughput is required.

Example:

System described by

Y(n) = y(n-1) + x(n) is IIR filter.

h(n)=an u(n) for n≥0