ANAND ENGINEERING COLLEGE, AGRA

DEPARTMENT OF ELECTRONICS &

INSTRUMENTATION

COURSE FILE (2010-2011)

MICROWAVE ENGINEERING
(VI SEM)

Prepared by: Shailendra Singh

Velocity Modulation Process
 Tutorial Plan
S. Topic Tutorial Ref.
No. No.
1 Rectangular Wave Guide: Field Components, 1
TE, TM Modes, Dominant TE10 mode, Field
Distribution, Power, Attenuation. Circular
Waveguides: TE, TM modes. Wave Velocities
2 Micro strip Transmission line (TL), Coupled TL, 2
Strip TL, Coupled Strip Line, Coplanar TL,
Microwave Cavities,
3 Micro strip Transmission line (TL), Coupled TL, 3
Strip TL, Coupled Strip Line, Coplanar TL,
Microwave Cavities,
4 Hybrid Couplers, Microwave Propagation in 4
ferrites, Faraday Rotation, Isolators, Circulators.
S parameter analysis of all components.
5 Microwave Tubes: Limitation of Conventional 5
Active Devices at Microwave frequency, Two
Cavity Klystron, Reflex Klystron
6 Magnetron, Traveling Wave Tube, Backward 6
Wave Oscillators: Their Schematic, Principle of
Operation, Performance Characteristic and their
applications.
7 Solid state amplifiers and oscillators: Microwave 7
Bipolar Transistor, Microwave tunnel diode,
Microwave Field-effect Transistor,
8 Transferred electron devices, Avalanche Transit 8
–time devices: IMPATT Diode, TRAPPAT
Diode,
9 Microwave Measurements: General set up of a 9
microwave test bench, Slotted line carriage,
VSWR Meter, microwave power measurements
techniques, Crystal Detector, frequency
measurement, wavelength measurements,
10 Impedance and Refection coefficient, VSWR, 10
Insertion and attenuation loss measurements,
measurement of antenna characteristics,
microwave link design.

Text Books:
Samuel Y. Liao, “Microwave Devices and Circuits”, 3rd Ed, Pearson Education.
A. Das and S. K. Das, “Microwave Engineering”, TMH.
 INDEX

1. PREAMBLE

2. SYLLABUS

3. TIME TABLE

4. LECTURE PLAN

5. TUTORIAL PLAN

6. LIST OF STUDENTS

7. NOTES

8. TUTORIALS

9. ASSIGNMENT

10. QUESTION BANK

11. BOOKS & REFRENCES

 PREAMBLE

Digital signal processing (DSP) is concerned with the representation of discrete time
signals by a sequence of numbers or symbols and the processing of these signals. Digital
signal processing and analog signal processing are subfields of signal processing.

The goal of DSP is usually to measure, filter and/or compress continuous real-world
analog signals. The first step is usually to convert the signal from an analog to a digital
form, by sampling it using an analog-to-digital converter (ADC), which turns the analog
signal into a stream of numbers. However, often, the required output signal is another
analog output signal, which requires a digital-to-analog converter (DAC). Even if this
process is more complex than analog processing and has a discrete value range, the
application of computational power to digital signal processing allows for many
advantages over analog processing in many applications, such as error detection and
correction in transmission as well as data compression.

DFT:-

In mathematics, the discrete Fourier transform (DFT) is a specific kind of discrete

transform, used in Fourier analysis. It transforms one function into another, which is
called the frequency domain representation, or simply the DFT, of the original function
(which is often a function in the time domain). But the DFT requires an input function
that is discrete and whose non-zero values have a limited (finite) duration. Such inputs
are often created by sampling a continuous function, like a person's voice. Unlike the
discrete-time Fourier transform (DTFT), it only evaluates enough frequency components
to reconstruct the finite segment that was analyzed. Using the DFT implies that the finite
segment that is analyzed is one period of an infinitely extended periodic signal; if this is
not actually true, a window function has to be used to reduce the artifacts in the spectrum.
For the same reason, the inverse DFT cannot reproduce the entire time domain, unless the
input happens to be periodic (forever). Therefore it is often said that the DFT is a
transform for Fourier analysis of finite-domain discrete-time functions. The sinusoidal
basis functions of the decomposition have the same properties.

The sequence of N complex numbers x0, ..., xN−1 is transformed into the sequence of N
complex numbers X0, ..., XN−1 by the DFT according to the formula:

where i is the imaginary unit and is a primitive N'th root of unity. (This expression
can also be written in terms of a DFT matrix; when scaled appropriately it becomes a
unitary matrix and the Xk can thus be viewed as coefficients of x in an orthonormal basis.)
The transform is sometimes denoted by the symbol , as in or or
.

FFT:-

A fast Fourier transform (FFT) is an efficient algorithm to compute the discrete

Fourier transform (DFT) and its inverse. There are many distinct FFT algorithms
involving a wide range of mathematics, from simple complex-number arithmetic to group
theory and number theory.

A DFT decomposes a sequence of values into components of different frequencies. This

operation is useful in many fields (see discrete Fourier transform for properties and
applications of the transform) but computing it directly from the definition is often too
slow to be practical. An FFT is a way to compute the same result more quickly:
computing a DFT of N points in the naive way, using the definition, takes O(N2)
arithmetical operations, while an FFT can compute the same result in only O(N log N)
operations. The difference in speed can be substantial, especially for long data sets where
N may be in the thousands or millions—in practice, the computation time can be reduced
by several orders of magnitude in such cases, and the improvement is roughly
proportional to N / log (N). This huge improvement made many DFT-based algorithms
practical; FFTs are of great importance to a wide variety of applications, from digital
signal processing and solving partial differential equations to algorithms for quick
multiplication of large integers.

The most well known FFT algorithms depend upon the factorization of N, but (contrary
to popular misconception) there are FFTs with O (N log N) complexity for all N, even for
prime N. Many FFT algorithms only depend on the fact that is an Nth primitive root of
unity, and thus can be applied to analogous transforms over any finite field, such as
number-theoretic transforms.

Since the inverse DFT is the same as the DFT, but with the opposite sign in the exponent
and a 1/N factor, any FFT algorithm can easily be adapted for it.

FIR:-

A finite impulse response (FIR) filter is a type of a signal processing filter whose
impulse response (or response to any finite length input) is of finite duration, because it
settles to zero in finite time. This is in contrast to infinite impulse response (IIR) filters,
which have internal feedback and may continue to respond indefinitely (usually
decaying). The impulse response of an Nth-order discrete-time FIR filter (i.e. with a
Kronecker delta impulse input) lasts for N+1 samples, and then dies to zero.

FIR filters can be discrete-time or continuous-time, and digital or analog.

IIR:-

Infinite impulse response (IIR) is a property of signal processing systems. Systems with
this property are known as IIR systems or, when dealing with filter systems, as IIR filters.
IIR systems have an impulse response function that is non-zero over an infinite length of
time. This is in contrast to finite impulse response (FIR) filters, which have fixed-
duration impulse responses. The simplest analog IIR filter is an RC filter made up of a
single resistor (R) feeding into a node shared with a single capacitor (C). This filter has
an exponential impulse response characterized by an RC time constant.

IIR filters may be implemented as either analog or digital filters. In digital IIR filters, the
output feedback is immediately apparent in the equations defining the output. Note that
unlike FIR filters, in designing IIR filters it is necessary to carefully consider the "time
zero" case in which the outputs of the filter have not yet been clearly defined.

Design of digital IIR filters is heavily dependent on that of their analog counterparts
because there are plenty of resources, works and straightforward design methods
concerning analog feedback filter design while there are hardly any for digital IIR filters.
As a result, usually, when a digital IIR filter is going to be implemented, an analog filter
(e.g. Chebyshev filter, Butterworth filter, Elliptic filter) is first designed and then is
converted to a digital filter by applying discretization techniques such as Bilinear
transform or Impulse invariance.

Example IIR filters include the Chebyshev filter, Butterworth filter, and the Bessel filter.