Q1. What is the difference between digital and analog signal?

An analog signal is a continuous signal that contains time-varying quantities. Unlike a digital
signal, which has a discrete value at each sampling point,an analog signal has constant
fluctuations. An analog pattern (represented as the curve) alongside a digital pattern
(represented as the discrete lines). Analog signals consist of infinite/uncountable number of
sample values. Where areas digital signals consist of finite sample values. It can be continuous
or discontinuous.

Q2. What is signal BW and channel BW?

The term Bandwidth is used to characterize both signals and channels. To understand
bandwidth,we need to know about signals. As a signal is composed of many elemental signals
with different frequencies, the difference between the highest of these frequencies and the
lowest one is said to be de bandwidth of this signal, in other words, the width occupied by the
signal in the frequency domain.
For a channel to be used to successfully transmit a signal, it must let pass
at least the bandwidth of said signal. Channels have limitations and usually will attenuate and
distort a signal outside of a certairange of frequencies (band). So, the term bandwidth is used
to refer to the characteristic of both signals and channels.
• Channel bandwidth is the width of the radio channel and so corresponds to the
spectrum that is assigned to be used by the system.

• Signal bandwidth is the actual width of the transmitted signal. It is often narrower than
the channel bandwidth to allow practical transmit filters to be used to ensure that the
modulated signal does not spread out beyond the channel bandwidth.
Q3. How we can get 3dB bandwidth?
Step1: -From the magnitude(dB) vs frequency(f) graph , we take the peak amplitude(dB) from the
curve. Let’s say it x dB.
Step2: - Subtract (x-3)dB from that peak amplitude.
Step3: -Take the corresponding frequencies (fh,fl) from the curve with respect to (x-3)dB point.
Step4: - Subtract fh-fl to get BW. BW=fh-fl.

Q4. Idea of Time domain and frequency domain. What is spectrum

analyzer?

The time-domain representation gives the amplitudes of the signal at the instants of time during
which it was sampled. However, in many cases you need to know the frequency content of a signal
rather than the amplitudes of the individual samples.

Fourier's theorem states that any waveform in the time domain can be represented by the weighted
sum of sines and cosines. The same waveform then can be represented in the frequency domain as
a pair of amplitude and phase values at each component frequency.
You can generate any waveform by adding sine waves, each with a particular amplitude and phase. The
following figure shows the original waveform, labeled sum, and its component frequencies. The
fundamental frequency is shown at the frequency f0, the second harmonic at frequency 2f0, and the
third harmonic at frequency 3f0.
In the frequency domain, you can separate conceptually the sine waves that add to form the complex
time-domain signal. The figure shows single frequency components, which spread out in the time
domain, as distinct impulses in the frequency domain. The amplitude of each frequency line is
the amplitude of the time waveform for that frequency component. The representation of a
signal in terms of its individual frequency components is the frequency- domain representation of
the signal. The frequency-domain representation might provide more insight about the signal and the
system from which it was generated.

A spectrum analyzer measures the magnitude of an input signal versus

frequency within the full frequency range of the instrument. The primary use is to measure the power
of the spectrum of known and unknown signals.
Q5. What is Inter Symbol Interference and how we can avoid it?
In telecommunication, intersymbol interference (ISI) is a form of distortion of a signal in which one
symbol interferes with subsequent symbols. This is an unwanted phenomenon as the previous
symbols have similar effect as noise, thus making the communication less reliable. The spreading of the
pulse beyond its allotted time interval causes it to interfere with neighboring pulses. ISI is usually
caused by (1) multipath propagation or the inherent linear or non-linear frequency response of a
communication channel, due channel filtering by RLC components, to causing successive symbols
to "blur" together, (2) Channel bandwidth less than Signal Bandwidth.
We have to follow “Nyquist Criteria for removal of ISI. That means sampling frequency should
higher or equal to the maximum frequency present in the signal.

Q6. What are the different types of noise affecting a signal and how we
can overcome it?
Atmospheric, cosmic and external noises: It is the noises that arise due to external atmospheric
disturbances like thunder, solar influences, industrial noises.
• White noise: White noise is the type of noise that affects all frequencies equally. It spreads up
from zero frequency upwards with a flat amplitude. It gains its name from the fact that
white light contains all colors, and hence frequencies equally, and white noise contains all
frequencies of interest equally.
• Flicker noise, 1/f noise: This type of noise occurs in almost all electronic devices. It has a variety
of causes, each related to the direct current flow. It has a frequency spectrum that falls
off steadily into the higher frequencies
• Phase noise: Phase noise is a form of RF noise that is visible on radio frequency, and other
signals. It appears in the form of phase jitter or perturbations on the signal. These
manifest themselves as sidebands that spread out either side of the signal or carrier.
• Shot noise: This form of noise that arises from the time-dependent fluctuations in electrical
current. This is caused by the discrete nature of electron charges. Shot noise is particularly
noticeable in semiconductor devices,like tunnel junctions, Schottky diodes and p-n junctions.
• Thermal noise: This form of noise, also referred to as Johnson or Johnson Nyquist noise arises
as a result of the thermal agitation of charge carriers - typically electrons - in a conductor. As
the temperature, and hence the agitation of the charge carriers increases so does the level of
noise.
 • Partition noise: Is a form of noise found in some circuits where the circuit is to divide in two
or more paths, like diode,BJT,etc.

❖ By efficient designing of filters,proper electromagnetic shielding of RF sensitive devices,

taking more samples of a signal and hence efficient reconstruction, analysis of
frequency spectrum to detect and remove unwanted harmonics, etc ,we can remove
some of the noises.

Q7. What is signal equalization and why it is needed and how it works?
In telecommunication, equalization is the reversal of distortion incurred by a signal transmitted
through a channel. Equalizers are used to render the frequency response—for instance of a
telephone line—flatfrom end-to-end.
Modern digital telephone systems have less trouble in the voice frequency range as only the local line
to the subscriber now remains in analog format, but DSL circuits operating in
the MHz range on those same wires may suffer severe attenuation distortion, which is dealt with by
automatic equalization or by abandoning the worst frequencies. In digital communications, the
equalizer's purpose is to reduce inter symbol interference (ISI) to allow recovery of the transmit
symbols. It may be a simple linear filter or a complex algorithm.
There are different equalization techniques like Zero Forcing Equalizer, Adaptive equalizer, etc.
The Zero-Forcing Equalizer applies the inverse of the channel frequency response to the received signal,
to restore the signal after the channel. It has many useful applications. For example, it is studied heavily
for IEEE 802.11n (MIMO) where knowing the channel allows recovery of the two or more streams
which will be received on top of each other on each antenna. The name Zero Forcing corresponds to
bringing down the inter symbol
interference (ISI) to zero in a noise free case. This will be useful when ISI is significant compared to noise.
For a channel with frequency response F(f) the zero forcing equalizer C(f) is constructed
by C(f)=1/F(f). Thus the combination of channel and equalizer gives a flat frequency response and linear
phase F(f)C(f)=1.

Q8. Difference between periodic and non-periodic signals

Periodic Signal
A signal is considered to be periodic signal when it is repeated over cycle of time or regular interval of
time. This means periodic signal repeats its pattern over a period. The function f(x) can be periodic if it
satisfies following equation,f(x + p) = f(x)
Example, the cosine signal is periodic with periodicity value of 2π.
Non-periodic Signal
A signal is considered to be non-periodic or aperiodic signal when it does not repeat its pattern over a
period (i.e. interval of time), Ex- Signal created by microphone or telephone when one or two words
are pronounced. In this application, composite signal cannot be periodic in nature.
Hence it is referred as aperiodic signal.
Q9. Difference between Fourier Series and Fourier transform.
The Fourier series is used to represent a periodic function by a discrete sum of complex
exponentials, while the Fourier transform is then used to represent a general,
nonperiodic/periodic function by a continuous superposition or integral of complex
exponentials. The Fourier transform can be viewed as the limit of the Fourier series of a function with
the period approaches to infinity, so the limits of integration change from one period to (−∞,∞).

In a classical approach it would not be possible to use the Fourier transform for a periodic function
which cannot be in L1(−∞,∞). The use of generalized functions, however, frees us of that restriction
and makes it possible to look at the Fourier transform of a periodic function. It can be shown that the
Fourier series coefficients of a periodic function are sampled values of the Fourier transform of one
period of the function.

Q10. Difference between Fourier and Laplace transform

Fourier is a subset of Laplace. Laplace is a more generalized transform. Fourier is used primarily for
steady state signal analysis, while Laplace is used for transient signal analysis. Laplace is good at looking
for the response to pulses, step functions, delta functions, while Fourier is good for continuous
signals.
The Laplace transform maps a function f(t) to a function F(s) of the complex variable s, where s=σ+jω.
If we set the real part of the complex variable s to zero, σ=0, the result is the Fourier transform F(jω)
which is essentially the frequency domain representation of f(t).
Therefore, Fourier transform is computed using imaginary axis jw, while Laplace transform is computed
using the full s plane.

Q11. Why do we use source encoding, line encoding and channel

encoding?
A source code is something which attempts to answer mathematically .The rough idea is, highly
probable symbols are assigned the shortest coded transmissions. Source coding can be of many types,
fixed length code, variable length code, etc. Goal of all source coding - using the channel (which can also
be a ‘time’ channel like a storage medium) most efficiently (transmit less, convey maximum).

Shannon once asked the question - “What is the maximum rate at which we can convey information
with small chance of error through this probabilistic channel?”, and thus was born channel coding,
which is the art of designing mapping the (source-coded) information
symbols to the noisy-channel such that communication happens at the maximum possible rate as well
as low error probabilities.
There are a number of useful line codes that we can use to help reduce our errors or to have other
positive effects on our signal. Line coding consists of representing the digital signal to be transported,
by an amplitude- and time-discrete signal, that is optimally tuned for the specific properties of the
physical channel (and of the receiving equipment). The waveform pattern of voltage or current used
to represent the 1s and 0s of a digital signal on a transmission link is called line encoding. The common
types of line encoding are UNRZ, UPRZ, BPRZ, BPNRZ, AMI, Manchester Coding Etc.
Efficient line coding is done to send clocking information as well as the message, sending as much
less DC component as possible.