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Poc Unit-5-1

This document discusses the digital representation of analog signals, emphasizing the necessity and advantages of digitization for clearer communication. It covers key concepts such as the sampling process, quantization, and Pulse Code Modulation (PCM), detailing the elements involved in digital communication systems. The document also explains the processes of sampling, quantization, and encoding, along with the importance of techniques like companding and error correction in digital signal transmission.
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0% found this document useful (0 votes)
3 views17 pages

Poc Unit-5-1

This document discusses the digital representation of analog signals, emphasizing the necessity and advantages of digitization for clearer communication. It covers key concepts such as the sampling process, quantization, and Pulse Code Modulation (PCM), detailing the elements involved in digital communication systems. The document also explains the processes of sampling, quantization, and encoding, along with the importance of techniques like companding and error correction in digital signal transmission.
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We take content rights seriously. If you suspect this is your content, claim it here.
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UNIT-5

Digital representation of analog signals:Introduction,Why Digitize Analog Sources?, The


Sampling process, Pulse Amplitude Modulation, Time Division Multiplexing, Pulse-Position
Modulation, Generation of PPM Waves, Detection of PPM Waves, The Quantization
Process, Quantization Noise, Pulse Code Modulation: Sampling, Quantization, Encoding,
Regeneration, Decoding, Filtering, Multiplexing

Introduction:

The communication that occurs in our day-to-day life is in the form of signals. These signals,
such as sound signals, generally, are analog in nature. When the communication needs to be
established over a distance, then the analog signals are sent through wire, using different
techniques for effective transmission.
Digitize Analog Sources?

The Necessity of Digitization:

The conventional methods of communication used analog signals for long distance
communications, which suffer from many losses such as distortion, interference, and other
losses including security breach.
In order to overcome these problems, the signals are digitized using different techniques. The
digitized signals allow the communication to be more clear and accurate without losses.
The following figure indicates the difference between analog and digital signals. The digital
signals consist of 1s and 0s which indicate High and Low values respectively.

Advantages of Digital Communication:

As the signals are digitized, there are many advantages of digital communication over analog
communication, such as −
 The effect of distortion, noise, and interference is much less in digital signals as they
are less affected.
 Digital circuits are more reliable.
 Digital circuits are easy to design and cheaper than analog circuits.
 The hardware implementation in digital circuits, is more flexible than analog.
 The occurrence of cross-talk is very rare in digital communication.
 The signal is un-altered as the pulse needs a high disturbance to alter its properties,
which is very difficult.
 Signal processing functions such as encryption and compression are employed in
digital circuits to maintain the secrecy of the information.
 The probability of error occurrence is reduced by employing error detecting and error
correcting codes.
 Spread spectrum technique is used to avoid signal jamming.
 Combining digital signals using Time Division Multiplexing TDMTDM is easier than
combining analog signals using Frequency Division Multiplexing FDMFDM.
 The configuring process of digital signals is easier than analog signals.
 Digital signals can be saved and retrieved more conveniently than analog signals.
 Many of the digital circuits have almost common encoding techniques and hence
similar devices can be used for a number of purposes.
 The capacity of the channel is effectively utilized by digital signals.

Elements of Digital Communication:

The elements which form a digital communication system is represented by the following
block diagram for the ease of understanding.

Following are the sections of the digital communication system.


Source
The source can be an analog signal. Example: A Sound signal
Input Transducer
This is a transducer which takes a physical input and converts it to an electrical signal
(Example: microphone). This block also consists of an analog to digital converter where a
digital signal is needed for further processes.
A digital signal is generally represented by a binary sequence.
Source Encoder
The source encoder compresses the data into minimum number of bits. This process helps in
effective utilization of the bandwidth. It removes the redundant
bits unnecessaryexcessbits,i.e.,zeroesunnecessaryexcessbits,i.e.,zeroes.
Channel Encoder
The channel encoder, does the coding for error correction. During the transmission of the
signal, due to the noise in the channel, the signal may get altered and hence to avoid this, the
channel encoder adds some redundant bits to the transmitted data. These are the error
correcting bits.
Digital Modulator
The signal to be transmitted is modulated here by a carrier. The signal is also converted to
analog from the digital sequence, in order to make it travel through the channel or medium.
Channel
The channel or a medium, allows the analog signal to transmit from the transmitter end to the
receiver end.
Digital Demodulator
This is the first step at the receiver end. The received signal is demodulated as well as
converted again from analog to digital. The signal gets reconstructed here.
Channel Decoder
The channel decoder, after detecting the sequence, does some error corrections. The
distortions which might occur during the transmission, are corrected by adding some
redundant bits. This addition of bits helps in the complete recovery of the original signal.
Source Decoder
The resultant signal is once again digitized by sampling and quantizing so that the pure
digital output is obtained without the loss of information. The source decoder recreates the
source output.
Output Transducer
This is the last block which converts the signal into the original physical form, which was at
the input of the transmitter. It converts the electrical signal into physical output (Example:
loud speaker).
Output Signal
This is the output which is produced after the whole process. Example − The sound signal
received.
This unit has dealt with the introduction, the digitization of signals, the advantages and the
elements of digital communications. In the coming chapters, we will learn about the concepts
of Digital communications, in detail.
The Sampling process:

The process of converting continuous time signals into equivalent discrete time signals, can
be termed as Sampling. A certain instant of data is continually sampled in the sampling
process.
The following figure shows a continuous-time signal x(t) and the corresponding sampled
signal xs(t). When x(t) is multiplied by a periodic impulse train, the sampled signal xs(t) is
obtained.

A sampling signal is a periodic train of pulses, having unit amplitude, sampled at equal
intervals of time Ts, which is called as sampling time. This data is transmitted at the time
instants Ts and the carrier signal is transmitted at the remaining time.
Sampling Rate
To discretize the signals, the gap between the samples should be fixed. That gap can be
termed as the sampling period Ts. Reciprocal of the sampling period is known as sampling
frequency or sampling rate fs.
Mathematically, we can write it as
fs=1Ts
Where,
fs is the sampling frequency or the sampling rate
Ts is the sampling period

Sampling Theorem

The sampling rate should be such that the data in the message signal should neither be lost
nor it should get over-lapped. The sampling theorem states that, “a signal can be exactly
reproduced if it is sampled at the rate fs, which is greater than or equal to twice the maximum
frequency of the given signal W.”
Mathematically, we can write it as
fs≥2W
Where,
 fs is the sampling rate
 W is the highest frequency of the given signal
If the sampling rate is equal to twice the maximum frequency of the given signal W, then it is
called as Nyquist rate.
The sampling theorem, which is also called as Nyquist theorem, delivers the theory of
sufficient sample rate in terms of bandwidth for the class of functions that are bandlimited.
For continuous-time signal x(t), which is band-limited in the frequency domain is represented
as shown in the following figure.

If the signal is sampled above Nyquist rate, then the original signal can be recovered. The
following figure explains a signal, if sampled at a higher rate than 2w in the frequency
domain.

If the same signal is sampled at a rate less than 2w, then the sampled signal would look like
the following figure.

We can observe from the above pattern that there is over-lapping of information, which leads
to mixing up and loss of information. This unwanted phenomenon of over-lapping is called
as Aliasing.
Aliasing can be referred to as “the phenomenon of a high-frequency component in the
spectrum of a signal, taking on the identity of a low-frequency component in the spectrum of
its sampled version.”
Hence, the sampling rate of the signal is chosen to be as Nyquist rate. If the sampling rate is
equal to twice the highest frequency of the given signal W, then the sampled signal would
look like the following figure.

In this case, the signal can be recovered without any loss. Hence, this is a good sampling rate.
Pulse Amplitude Modulation:
The amplitude of the pulses of the carrier pulse train is varied in accordance with the
modulating signal, that is amplitude of the pulses depends on the value of m(t) during the
time of pulse.

• In fact the pulses in a PAM signal may of Flat-top type or natural type or
ideal type.
• The Flat-top PAM is most popular and is widely used. The reason for
using Flat-top PAM is that during the transmission, the noise interferes
with the top of the transmitted pulses and this noise can be easily removed
if the PAM pulse as Flat-top.
• In natural samples PAM signal, the pulse has varying top in accordance
with the signal variation. Such type of pulse is received at the receiver, it
is always contaminated by noise. Then it becomes quite difficult to
determine the shape of the top of the pulse and thus amplitude detection of
the pulse is not exact.
Generation of PAM

• There are two operations involved in the generation of PAM signal


1. Instantaneous sampling of the message signal m(t) every Ts seconds, where
the sampling rate fs = 1/Ts is chosen in accordance with the sampling theorem.
2. Lengthening the duration of each sample so obtained to some constant value T.

Quantizing processor:

The analog-to-digital converters perform this type of function to create a series of digital
values out of the given analog signal. The following figure represents an analog signal. This
signal to get converted into digital, has to undergo sampling and quantizing.
The quantizing of an analog signal is done by discretizing the signal with a number of
quantization levels. Quantization is representing the sampled values of the amplitude by a
finite set of levels, which means converting a continuous-amplitude sample into a discrete-
time signal.
The following figure shows how an analog signal gets quantized. The blue line represents
analog signal while the brown one represents the quantized signal.

Both sampling and quantization result in the loss of information. The quality of a Quantizer
output depends upon the number of quantization levels used. The discrete amplitudes of the
quantized output are called as representation levels or reconstruction levels. The spacing
between the two adjacent representation levels is called a quantum or step-size.
The following figure shows the resultant quantized signal which is the digital form for the
given analog signal.

This is also called as Stair-case waveform, in accordance with its shape.

Types of Quantization:
There are two types of Quantization - Uniform Quantization and Non-uniform Quantization.
The type of quantization in which the quantization levels are uniformly spaced is termed as
a Uniform Quantization. The type of quantization in which the quantization levels are
unequal and mostly the relation between them is logarithmic, is termed as a Non-uniform
Quantization.
There are two types of uniform quantization. They are Mid-Rise type and Mid-Tread type.
The following figures represent the two types of uniform quantization.

Figure 1 shows the mid-rise type and figure 2 shows the mid-tread type of uniform
quantization.
 The Mid-Rise type is so called because the origin lies in the middle of a raising part of
the stair-case like graph. The quantization levels in this type are even in number.
 The Mid-tread type is so called because the origin lies in the middle of a tread of the
stair-case like graph. The quantization levels in this type are odd in number.
 Both the mid-rise and mid-tread type of uniform quantizers are symmetric about the
origin.

Quantization Error:

For any system, during its functioning, there is always a difference in the values of its input
and output. The processing of the system results in an error, which is the difference of those
values.
The difference between an input value and its quantized value is called a Quantization
Error. A Quantizer is a logarithmic function that performs Quantization rounding off the
value rounding off the value. An analog-to-digital converter (ADC) works as a quantizer.
The following figure illustrates an example for a quantization error, indicating the difference
between the original signal and the quantized signal.
Quantization Noise:
It is a type of quantization error, which usually occurs in analog audio signal, while
quantizing it to digital. For example, in music, the signals keep changing continuously, where
a regularity is not found in errors. Such errors create a wideband noise called
as Quantization Noise.

Companding in PCM:

The word Companding is a combination of Compressing and Expanding, which means that
it does both. This is a non-linear technique used in PCM which compresses the data at the
transmitter and expands the same data at the receiver. The effects of noise and crosstalk are
reduced by using this technique.
There are two types of Companding techniques. They are −
A-law Companding Technique
 Uniform quantization is achieved at A = 1, where the characteristic curve is linear and
no compression is done.
 A-law has mid-rise at the origin. Hence, it contains a non-zero value.
 A-law companding is used for PCM telephone systems.
µ-law Companding Technique
 Uniform quantization is achieved at µ = 0, where the characteristic curve is linear and
no compression is done.
 µ-law has mid-tread at the origin. Hence, it contains a zero value.
 µ-law companding is used for speech and music signals.
µ-law is used in North America and Japan.
Pulse Code Modulation:
A signal is pulse code modulated to convert its analog information into a binary sequence,
i.e., 1s and 0s. The output of a PCM will resemble a binary sequence. The following figure
shows an example of PCM output with respect to instantaneous values of a given sine wave.
Instead of a pulse train, PCM produces a series of numbers or digits, and hence this process is
called as digital. Each one of these digits, though in binary code, represent the approximate
amplitude of the signal sample at that instant.
In Pulse Code Modulation, the message signal is represented by a sequence of coded pulses.
This message signal is achieved by representing the signal in discrete form in both time and
amplitude.

Basic Elements of PCM

The transmitter section of a Pulse Code Modulator circuit consists of Sampling,


Quantizing and Encoding, which are performed in the analog-to-digital converter section.
The low pass filter prior to sampling prevents aliasing of the message signal.
The basic operations in the receiver section are regeneration of impaired signals,
decoding, and reconstruction of the quantized pulse train. Following is the block diagram of
PCM which represents the basic elements of both the transmitter and the receiver sections.
Low Pass Filter:

This filter eliminates the high frequency components present in the input analog signal which
is greater than the highest frequency of the message signal, to avoid aliasing of the message
signal.
Sampler
This is the technique which helps to collect the sample data at instantaneous values of
message signal, so as to reconstruct the original signal. The sampling rate must be greater
than twice the highest frequency component W of the message signal, in accordance with the
sampling theorem.
Quantizer:
Quantizing is a process of reducing the excessive bits and confining the data. The sampled
output when given to Quantizer reduces the redundant bits and compresses the value.
Encoder:

The digitization of analog signal is done by the encoder. It designates each quantized level by
a binary code. The sampling done here is the sample-and-hold process. These three
sections LPF,Sampler,and QuantizerLPF,Sampler,andQuantizer will act as an analog to
digital converter. Encoding minimizes the bandwidth used.

Regenerative Repeater:
This section increases the signal strength. The output of the channel also has one regenerative
repeater circuit, to compensate the signal loss and reconstruct the signal, and also to increase
its strength.
Decoder:
The decoder circuit decodes the pulse coded waveform to reproduce the original signal. This
circuit acts as the demodulator.
Reconstruction Filter:
After the digital-to-analog conversion is done by the regenerative circuit and the decoder, a
low-pass filter is employed, called as the reconstruction filter to get back the original signal.
Hence, the Pulse Code Modulator circuit digitizes the given analog signal, codes it and
samples it, and then transmits it in an analog form. This whole process is repeated in a
reverse pattern to obtain the original signal.
PCM Transmitter

The detailed diagram of a practical PCM transmitter is shown below


 LPF: Here, the message signal which is in the continuous time form, is allowed to pass
through a low pass filter (LPF). This LPF whose cutoff frequency is fm eliminates the
high-frequency components of the signal and passes only the frequency components
that lie below fm.
 Sampler: The output of the LPF is then fed to a sampler where the analog input signal
is sampled at regular intervals. The sampling of the signal is done at the rate of fs. This
sampling frequency is so selected that it must follow the sampling theorem that is
expressed as:
fs ≥ 2fm

The output of the sampler is a signal that is discrete time continuous amplitude signal denoted
as nTs which is nothing but a PAM signal.

 Quantizer: A quantizer is a unit that rounds off each sample to the nearest discrete
level. The sampler provides a continuous range signal and hence still an analog one.
The quantizer performs the approximation of each sample thus assigning it a particular
discrete level.
As it basically rounds off the value to a certain level this shows some variation by the actual
amount. Thus we can say, quantizing a signal introduces some distortion or noise into it. This
is known as quantization error.

This noise factor is somewhat better than the channel noise as it is controllable.

A quantizer can be of two types, uniform and non-uniform quantizer. In uniform


quantizer, there exists a uniform spacing in between the level. As against, in non-uniform
quantizer, the spacing in between the levels is not uniform. Here, we have employed a
uniform quantizer.

For a low signal level, the quantization error is high i.e., bad SNR. But, for a high signal
level, the quantization error is low providing good SNR.

The figure below shows the sampling of analog signal and further quantization of the samples
 Encoder: An encoder performs the conversion of the quantized signal into binary
codes. This unit generates a digitally encoded signal which is a sequence of binary
pulses that acts as the modulated output.
As it is a binary encoder thus generates a binary coded sequence. That is transmitted
through the transmission path.

Transmission path in a PCM system

A PCM system has a better control over signal distortion introduced during transmission
through the channel than other systems. PCM achieves low signal distortion by employing
regenerative receivers along the transmission path.

The channel introduces distortion in the signal during transmission. This distortion is
eliminated by the regenerator in order to provide a distortionless PCM signal. Resultantly,
enhancing the transmission ability of the system.

The figure below shows the block diagram of a regenerative repeater. It basically performs
equalization and timing and then executes decision making.
The PCM signal when provided to the regenerative repeater, the equalizer circuit at the
beginning performs the reshaping of the distorted signal. At the same time, the timing circuit
generates a pulse train that is a derivative of input PCM pulses.

This pulse train is then utilized by the decision-making device in order to sample the PCM
pulses. This sampling is done at the instant where maximum SNR can be achieved. In this
way, the decision-making device generates the distortionless PCM wave.

PCM Receiver:

The figure below shows the functional block diagram of a PCM receiver

 Regenerator: A regenerative repeater is placed at the receiving end also so as to have


an exact PCM transmitted signal. Here, also the regenerator works in a similar manner
as that when employed in the transmission path. It eliminates the channel induced noise
and reshapes the pulse.
 DAC and Sampler: Digital to analog converter performs the conversion of digital
signal again into its analog form by making use of the sampler. As the actual message
signal was analog thus at the receiver end there is a necessity to again convert it into its
original form.
 LPF: The sampler generates analog signal but that is not the original message signal.
Thus, the output of the sampler is fed to the LPF having cutoff frequency fm. This is
sometimes termed as the reconstruction filter that produces the original message signal.
The process done at the transmitter is somewhat reversed at the receiver in order to generate
the original analog message signal. The figure below shows the reconstruction of the actual
analog message signal at the receiver.
Transmission bandwidth in Pulse Code Modulation:

The transmission bandwidth of a PCM system is associated with a number of bits per
sample. If the number of bits per sample increases, the bandwidth also increases. In order to
have a good approximation, a large number of levels must be used but that will lead to a
larger bandwidth requirement.

Let us consider each quantizer level is represented by ‘n’ binary digits. Then the levels
represented by n binary digits is given as,

q = 2n

: q is the digital level of the quantizer

Every sample is changed into n bits, thus, a number of bit per sample is ‘n’.

As we have already discussed the number of samples per second is fs. Hence the number of
bits per second which is also termed as signalling rate is given as,

r = n fs

As transmission bandwidth is half the signalling rate, hence

As we have already discussed, r = nfs

Therefore,

But we know, fs ≥ 2fm

Thus the bandwidth of the PCM system is given as,

BW ≥ n fm

Advantages of PCM

1. Immune to channel induced noise and distortion.


2. Repeaters can be employed along the transmitting channel.
3. Encoders allow secured data transmission.
4. It ensures uniform transmission quality.
Disadvantages of PCM

1. Pulse code modulation increases the transmission bandwidth.


2. A PCM system is somewhat more complex than another system.
Thus from the above discussion, we can conclude that a PCM system, transmits data in a
coded format, that ensures secured transmission. But, this at the same time needs decoding
system in order to reproduce exact message signal that increases system complexity.

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