Digital Communication
Digital Communication
In analog communication, the message signals are continuous-time and can take infinite
amplitude levels. When these signals are transmitted over long distances, even a small
disturbance (noise) can cause distortion in the signal. The signal gets distorted, and it is difficult
to recover the original signal in the presence of noise at the receiver. In digital communication,
errors in the transmission can be detected and corrected using error correction coding
techniques. Hence digital communication has more benefits as compared to analog
communication. In digital communication, the message signal is in a discrete form with finite
amplitude levels. If the message signal is analog, it must be converted to the digital format by
Analog-to-Digital Conversion.
The signal must be converted to digital form for any analog information to be transmitted using
a digital communication system. The analog-to-digital conversion process involves sampling,
quantizing, and encoding the analog signal. The digitized signal is then modulated using digital
modulation techniques.
Learning Outcomes:
The basic principles of Analog to Digital Conversion are shown in Fig 8.1.1. The analog signals
are converted into digital signals with simple steps called sampling, quantizing, and encoding.
Sampling can be observed in numerous real-life applications. For example, music CDs
(Compact Discs) are produced by sampling live sound at frequent intervals and then quantizing
and encoding each sample. If the sampling rate is too high, the human sensory organs cannot
discriminate each frame they are played back. If the sampling rate is low, there will be
distortion in the reconstructed picture obtained from the digitized samples. Therefore, while
sampling an analog signal, there is a minimum sampling rate requirement, called the Nyquist
Sampling rate, to avoid distortion in the reconstructed signal. The sampling theorem states, "It
is possible to reconstruct a band-limited analog signal from periodic samples, as long as the
sampling rate is at least twice the signal's highest frequency component.”
Where fs is the sampling frequency, and fc is the highest frequency contained in the signal. This
theorem is also commonly called the Nyquist sampling theorem .
If a voice signal has frequencies ranging from 0 to 4kHz (Low pass signal), then according to
the Nyquist Sampling Theorem, to sample this signal without distortion, the minimum required
sampling rate is equal to 8kHz. If an analog signal has frequency components ranging from 2
kHz to 5 kHz (Band pass signal), then according to the sampling theorem, the Nyquist sampling
rate is equal to twice that of the signal's bandwidth. ie. 2*(5-2) kHz = 6 kHz and not 10 kHz.
Self -test:
6 kHz
12 kHz
14.4 kHz
20 kHz
Example Problem:
1 Consider the analog signal x (t) =3cos100 π t. Determine the minimum sampling
rate required to avoid aliasing.
Solution:
The frequency of the analog signal can be calculated as 2πfc=100π.
Therefore fc=50Hz. According to the Nyquist sampling rate, the minimum sampling
rate required to avoid aliasing is fs = 100 Hz.
Exercise:
1 Consider the analog signal x (t) = 3cos50πt+10sin300πt -cos100πt. What is
the Nyquist rate of sampling for this signal? (Ans: 300Hz)
Note: The effect of incorrect sampling rate can be seen when the rotation of a helicopter blade is
observed. As the speed of the blade increases, our eyes are under sampling the true speed of the
blade with a rate which is limited by the human brain. Similarly, in movies, when the motion of car
wheels with increasing speed is observed, the movie camera is under sampling the motion of car
wheels by sampling at a rate equal to the fixed frames per second of the camera. In both the
examples it is observed that as the speed increases, it creates an illusion of backward rotation.
This is because in both cases the actual speed is under sampled.
For Analogy of sampling to Wagon wheel effect, visit the following link:
http://www.youtube.com/watch?v=6XwgbHjRo30
A pulse train has three parameters: Pulse Amplitude, Pulse Width, and the instant of
occurrence of the pulse – Pulse Position. The information to be transmitted can be varied
any of these parameters according to the instantaneous amplitude of the modulating signal.
Three different pulse modulation types: Pulse Amplitude Modulation, Pulse Width
Modulation, and Pulse Position Modulation, as shown in Fig 8.1.2.
In this method, both the amplitude and the duration are .kept constant while the position of
each pulse in relation to the position of a recurrent reference pulse is varied by each
instantaneous sampled value of the modulating signal. (Refer to Fig 8.1.2) PPM is used in
both analog and digital data transmission. It is commonly used in optical fiber
communication, remote controls for TV, toys, etc.
The pulse modulation techniques discussed in the previous section are used to transmit
message signals over short distances. They are also called baseband modulation
techniques. Suppose message signals have to be transmitted over longer distances. In that
case, the pulse modulation technique is unsuitable because the modulated signal is in
digital form (information is contained in either the amplitude, width, or position of the
train of pulses). For this reason, digital modulation techniques are also called bandpass
modulation techniques. Here a continuous signal such as a high-frequency sinusoid acts
like a carrier.
ELEMENTS OF DIGITAL COMMUNICATION SYSTEMS:
Fig 8.1.4 shows the functional elements of a digital communication system. The function
of each block is explained as follows:
Many of the real-world signals are physical in nature. An input transducer is a device used
to convert these physical parameters to corresponding electrical signals. The input
transducer converts voice, speech, music, or image to electrical signals. Examples of input
transducers are microphones, cameras, etc. Usually, the output signal from the transducer
will be analog in nature. This analog signal is converted into digital form using an analog-
to-digital converter. The analog-to-digital conversion consists of sampling, quantizing, and
encoding. In the case of the output data of a computer, the signal is available in digital form
directly.
The source coding aims to efficiently represent the digital signal with as much fewer bits
as possible. This will reduce the bandwidth required for transmission—Ex: Huffman
coding. The source decoder performs the inverse operation of the source encoder. ie. It is
used to get back the data in its original representation.
c. Channel Encoder / Decoder
Channel coding consists of systematically adding extra bits in a known manner to the digital
data to be transmitted. These extra bits do not convey any information but help the receiver
detect and correct some errors in the received data. Channel encoding is done by using
either Block Coding or Convolution Coding methods. The channel decoder performs the
inverse operation of the channel encoder. It extracts the digital data from its encoded form
with minimum possible error. The decoder helps detect and correct errors in the received
data that gets introduced during transmission.
d. Modulator/ Demodulator
The Modulator converts the digital input information into an electrical waveform suitable
for transmission over the communication channel. Mainly there are three types of Digital
modulation techniques viz., Amplitude Shift Keying (ASK), Frequency Shift Keying
(FSK), and Phase Shift Keying (PSK). The demodulator accomplishes the extraction of the
digital data from the received signal.
e. Channel
The channels are either wired, coaxial cable, optical fiber, or wireless (free space), such as
a radio channel, satellite channel, or a combination of these. The communication channels
have only finite bandwidth, and the signal often suffers amplitude and phase distortion as
it travels over the channel. The signal power in the channel decreases with distance. It may
also get corrupted by unwanted, unpredictable electrical signals called noise. The two
important parameters used to measure the channel characteristics are Signal to Noise power
Ratio (SNR) and usable bandwidth.
There are three basic types of digital modulation techniques. They are:
i. Amplitude Shift Keying (ASK)
ii. Frequency Shift Keying (FSK)
iii. Phase Shift Keying (PSK)
In all these techniques, the amplitude, frequency or phase of a sinusoidal carrier varies to
represent the information to be sent. The digitized data is mapped into the above three
aspects of the sine wave and transmitted. The sine wave at the receiver is remapped back
to the information. The digital modulation techniques are widely used in MODEMs (
MOdulator DEModulator), mobile communication etc. Usually, FSK and PSK modulations
are more frequently used than ASK.
Summary