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Module 2 ppt

The document discusses the principles and advantages of Digital Communication Techniques (DCT), highlighting the robustness against noise and the use of regenerative repeaters. It explains the process of converting analog signals to digital using various modulation and encoding techniques, including Pulse Code Modulation (PCM) and Differential Pulse Code Modulation (DPCM). Additionally, it covers the challenges such as increased channel bandwidth and circuit complexity, while detailing the components involved in digital communication systems.

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Dipanjan Dutta
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© © All Rights Reserved
Available Formats
Download as PDF, TXT or read online on Scribd
0% found this document useful (0 votes)
4 views

Module 2 ppt

The document discusses the principles and advantages of Digital Communication Techniques (DCT), highlighting the robustness against noise and the use of regenerative repeaters. It explains the process of converting analog signals to digital using various modulation and encoding techniques, including Pulse Code Modulation (PCM) and Differential Pulse Code Modulation (DPCM). Additionally, it covers the challenges such as increased channel bandwidth and circuit complexity, while detailing the components involved in digital communication systems.

Uploaded by

Dipanjan Dutta
Copyright
© © All Rights Reserved
Available Formats
Download as PDF, TXT or read online on Scribd
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• Modulating signal is digital

• Carrier signal in analog

m(t)

c(t)

S(t)
Advantages of DCT
• Ruggedness to transmission noise and interference
The recovery of the message signal in a digital system is much easier than the analog
system. As the receiver has only to decide the presence or absence of the pulse.

m(t) Sampler Quantizer Encoder M(n)

• Use of Regenerative repeaters along the transmission line


• These are devices which remove noise, regenerates the original signal and re-
transmits them. Thus, avoiding accumulation of noise.
• Common format can be used for transmitting different types of signal like
speech, video, computer data, etc. and this is called line coding.

• Improved security of the transmitting signal due to Data Encryption


Disadvantages of DCT
• Increased Channel Bandwidth
• This requirement is fulfilled by the availability of wideband channels like fiber
optics and satellite channels.
• Increased Circuit Complexity & Cost
• This requirement is being addressed by the availability of VLSI techniques which
makes it cost effective.

• Channel Bandwidth: It is the range of frequencies a channel can transmit


with reasonable fidelity.
• Message Bandwidth: It is the measure of extent of significant spectral
content in a message signal.
• Lowpass Filter Signal: It is a signal whose significant spectral content is
centered about the origin.
• Bandpass Filter Signal: It is a signal whose significant spectral content is
centered about the center frequency (fc)
Block Diagram of Digital Communication System
Block wise Explanation
• Digital Source: The analog message signal is passed through a sampler,
followed by a quantizer to obtain a digital source.
• Source Encoder: It maps the incoming digital signal into another signal in
digital domain. This process gives an effective representation of source
output by removing certain redundant bits from the source output. This
helps in reduction of transmission bandwidth. i.e. Hoffman Coding – It is a
type of very least bit code (VLC) which uses lesser number of coding.
• Channel Encoder: It maps the incoming digital signal from the source
encoder, into the channel input to reduce noise effects. This is done by
introduction of some redundant bits in a prescribed manner which helps in
error detection and error correction of the transmitted bits. The channel
encoder and decoder help in reliable communication over the channel.
• Types of channel coding: Block Coding
Convolution Coding
• Modulation: It is needed for efficient transmission of he signal over the
channel.
• Need for Modulation:
• Practicability of Antenna size

• Multiplexing

• Reduction of Noise

• Narrow Banding

• In digital communication, modulation involves the process


of shifting or it is also known as keying the amplitude,
frequency or phase of the sinusoidal carrier signal according
to the incoming message data from the channel encoder.
• Choice of Modulation Technique depends on:
• Maximum data rate
• Minimum probability of symbol error
• Minimum transmitted power rate
• Minimum channel bandwidth
• Maximum resistance to noise & interference
• Minimum circuit complexity
• Channels are:
• Point to point
• Broadcast
• Fibre optics
• Satellite
Digital Coding of Analog Signals
• For transmission of Analog message signals, audio or video signals through
digital means, we need to first convert them into digital signal. The process
of Analog to Digital conversion is called Digital Pulse Modulation (DPM).
The term pulse modulation is justified in the first process of ADC involving
the process called sampling, where analog message signal is represented
by uniformly spaced pulses, whose amplitude is modulated.

• Process of ADC
• Pulse Code Modulation (PCM)
• Differential PCM (DPCM)
• Delta Modulation (DM)
• Adaptive Delta Modulation (ADM)

*These techniques are not used as the conventional techniques in


modulation scenario but are used mainly as methods of analog to digital
conversion
PCM Block Diagram
Block wise Explanation
• Low Pass Filter: It is used as an pre-aliasing filter. The information signal is
band limited to a maximum frequency of fmmax. The unnecessary frequency
components are required to be removed and there are no aliasing effect
for proper sampling process.
• Sampler: It is a process by which a continuous time signal is represented
by a sequence of pulses at periodic instances of time.
• Sampling Techniques:
• Instantaneous Sampling
• Natural Sampling
• Flat-top Sampling
Flat-top sampling is preferred because the sampling rate (fs) must be grater than
the maximum frequency present in the analog signal, that satisfies the Nyquist
rate [𝑓𝑠 ≥ 2𝑓𝑚𝑚𝑎𝑥 ]
• Quantizer: It is a process in which amplitudes of the sampled values are
rounded off to the nearest one of the finite set of discrete levels, which are
also known as quantized levels, so that the signal becomes discrete both in
amplitude and time.
• Quantization Process:
Consider an analog signal m(t) with minimum amplitude ‘VL’ and
maximum amplitude ‘VH’. The whole range 𝑉𝐿 → 𝑉𝐻 is divided into ‘L’ equal
intervals with step size:
𝑉𝐻 − 𝑉𝐿
𝑆=
𝐿
The number of intervals are calculated
according to the relation
𝐿 = 2𝑛
where, ‘n’ is the number of encoded bits.
Whenever the analog signal m(t) lies in that
range, the quantized signal mq(t) maintains a
constant level m0 which is the quantization
level located at the midpoint of the step.
Similarly, when m(t) is in the range L1, mq(t)
maintains a constant level at L1. Thus, the
quantized signal mq(t) takes value from m0 to
m7. The difference between output quantized
signal and the input analog signal m(t) gives
rise to an error known as Quantization Error
(Qe). The error can be reduced by decreasing
the step size and by increasing the number of
quantization levels.
• Two types of Quantization Process
• Uniform Quantization: (Step size is constant)

The maximum quantization error in this quantization process is given by


𝑆
𝑄𝑒𝑚𝑎𝑥 =
2
If VH=mP & VL=-mP,
The step-size (S)
𝑚𝑝 −(−𝑚𝑝 ) 2𝑚𝑝
𝑆= = for sine wave
𝐿 𝐿
• Non-Uniform Quantization: (Step size changes according to input signal)
In non-uniform quantization, the step size changes according to input
signal. For small amplitude signals (SNR0) reduces but it is desirable that
it remains constant for all input ranges. The quantization noise power,
𝑆2
𝑁𝑞 = , 𝑤ℎ𝑒𝑟𝑒 𝑆 → 𝑠𝑡𝑒𝑝 𝑠𝑖𝑧𝑒 𝑖𝑛 𝑣𝑜𝑙𝑡𝑠
12
Thus, by reducing ‘S’ for weak signals we can reduce quantization noise
power and thereby improving SNR.

This can be implemented by a


process called COMPANDER which is
equal to Compressor + Expander.
Here, the normalized input signal ‘m’ is input samples
and ‘mP’ is the peak amplitude of input samples.

• The compressor maps the input increments in ‘Δm’ to larger


output increments ‘Δy’ for small amplitude signals and vice-
versa for large amplitude signals.

• The interval ‘Δm’ thus has larger steps or levels and hence,
smaller ‘Δy’ step size, thereby reducing the quantization noise
power and improving the SNR for low amplitude signals.

Uniform O/P
I/P Compressor Expander
Quantizer
• The output of the compressor is uniform, hence non-uniform
quantization can be implemented using uniform quantization,
which makes the circuit implementation less complex and less
expensive.
• A simple semiconductor diode can be used to realize the
compressor whose V-I characteristics can be given as:

𝐾𝑇 𝐼
𝑉= ln(1 + )
𝑞 𝐼𝑆

• Expander:
• The compressor input samples are restored back with the
original I/P values with the expander, thus the characteristics
of Expander is just the complementary of compressor.
Expander is basically a receiving device.
Encoder

• The encoder forms the codeword and translates them into appropriate PCM
waveform for transmission through the channel.

• CODE: particular arrangement of discrete events is called a code.


• SYMBOL: one such discrete event in a code is called a symbol.

• Properties of a Line Coder:


• It should require less transmission bandwidth

• It should have less bit error rate

• It should contain adequate amount of timing pulses to extract pulses for synchronization

• It should require low transmission power


• Different types of Line Coding formats:

Can be represented as
i. Unipolar Format

ii. Polar Format NRZ

iii. Bipolar Format

iv. Manchester Format RZ

v. M-ary Format
(Polar Quaternary Signal)

NRZ: If the pulse occupies complete duration of a symbol, it is Non-Return to Zero.

RZ: If the pulse occupies half the duration of a symbol, it is Return to Zero.
• Unipolar Format:
• Symbol ‘1’ is represented by a transmitting pulse
• Symbol ‘0’ is represented by switching OFF the pulse
• Polar Format:
• Symbol ‘1’ is represented by a positive pulse
• Symbol ‘0’ is represented by a negative pulse
• Bipolar Format:
• Symbol ‘1’ is represented by a positive pulse, followed by a negative pulse for alternate
occurrences
• Symbol ‘0’ is represented by switching OFF the pulse
• Manchester Format:
• Symbol ‘1’ is represented by a positive pulse for half the duration and negative pulse for
the next half.
• Symbol ‘0’ is represented by a negative pulse for half the duration and positive pulse for
the next half.

N
• M-ary Format:
• In this coding ‘M’ is the number of possible output voltage levels far a given
number of encoding bits ‘n’. The relation is given as:
𝑀 = 2𝑛
• In polar quaternary signaling, 2-bit called di-bits are transmitted at a time.
• For this, two types of coding scheme available with 𝑛 = 2.

Natural Coding Grey Coding

Easy to generate and also can withstand the effect of Each output level differs by only 1 bit. This property helps
noise. in error detecting and error correcting. This is mostly
used in digital communication.
Regenerative Repeater
Distorted PCM Decision Clean New pulse
Amplitude
Waveform Making Of PCM
Equalizer
Device

Timing Circuit

• Amplitude Equalizer: It removes any amplitude distortion and phase distortion


from the received distorted PCM signal.
• Timing Circuit: It generates timing signals or pulses from the equalized pulses
at the instance of time, when SNR is high. These timing pulses is then given to
the decision device to enable it.
• Decision Making Device: The timing pulses received from the timing circuit,
activates the decision device, when SNR is high. The decision device takes the
decision of generating a clean pulse, according to a defined threshold level.
Receiver

• Decoder: The first operation is to regenerate the received pulses.


These clean pulses are then regrouped into code words and
decoded into a quantized PAM signal. This decoding process
involves, generating a pulse, amplitude of which is a linear sum of
all the pulses in the code word; with each pulse weighted by its
place value in a code.

• Reconstruction Filter: A low pass filter is generally used as a re-


constructive filter to get back the analog signal. The cut-off
frequency of this low pass filter is said to be ‘X’Hz, which is the
frequency of the input analog signal, and LPF is basically an
integrator, used as a linear wave shaping circuit.
Que: The information in an analog signal is to be transmitted over a PCM
system with an accuracy of ±0.1%. The analog voltage waveform has a max
frequency of 100Hz and the amplitude ranges from -10V to +10V. Determine:
a) Minimum sampling rate
b) No.of bits in the PCM system
c) Minimum bit-rate required
d) Minimum channel bandwidth required
For the transmission.
DPCM [Differential Pulse Code Modulation]
• Advantages: Lesser transmission bandwidth and improvement in
SNR where as PCM requires a larger transmission bandwidth.

• The basic concept in PCM is when we sample a signal at sampling


rate 𝑓𝑠 > 2𝑓𝑚 , we find that the sequence of samples are highly
corelated meaning, the signal is not rapidly changing from one
sample to the next sample. Therefore, the variance of the
difference between adjacent samples is less than the variance of
the input samples.

• If we know the behavior of the past sample to a certain point of


time, we can predict the next sample value. Since, the samples
generated are highly corelated, these contain a lot of redundant
information. By removing these redundant information, we can
get a more efficiently coded signal.
• In DPCM technique, the difference between unquantized input and its
predicted value is encoded and transmitted.
• The difference is called Prediction Error, denoted as :
𝑒 𝑛𝑇𝑆 = 𝑥 𝑛𝑇𝑆 − 𝑥ො 𝑛𝑇𝑆
𝑥 𝑛𝑇𝑆 : is unquantized input sample value, generated when signal 𝑥 𝑡 is
sampled at Nyquist rate.
𝑥ො 𝑛𝑇𝑆 : is the predicted value of the input sample which is generated from the
predictor.
DPCM Transmitter

𝑥 𝑛𝑇𝑆 𝑒 𝑛𝑇𝑆 𝑣 𝑛𝑇𝑆 𝑏 𝑛𝑇𝑆


Ʃ 𝑄𝑢𝑎𝑛𝑡𝑖𝑧𝑒𝑟 𝐸𝑛𝑐𝑜𝑑𝑒𝑟

𝑥ො 𝑛𝑇𝑆
Ʃ

𝑃𝑟𝑒𝑑𝑖𝑐𝑡𝑜𝑟 𝑢 𝑛𝑇𝑆
𝑒 𝑛𝑇𝑆 : Prediction error 𝑒 𝑛𝑇𝑆 = 𝑥 𝑛𝑇𝑆 − 𝑥ො 𝑛𝑇𝑆 … (1)

𝑥 𝑛𝑇𝑆 : Unsampled input signal

𝑥ො 𝑛𝑇𝑆 : Predicted value of input signal

𝑣 𝑛𝑇𝑆 : Output of the quantizer 𝑣 𝑛𝑇𝑆 = 𝑒 𝑛𝑇𝑆 + 𝑞 𝑛𝑇𝑆 …(2)

𝑢 𝑛𝑇𝑆 : Input to the predictor

𝑞 𝑛𝑇𝑆 : Quantization error

𝑏 𝑛𝑇𝑆 : DPCM output signal

𝑢 𝑛𝑇𝑆 = 𝑣 𝑛𝑇𝑆 + 𝑥ො 𝑛𝑇𝑆 …(3)


From eq(2)
= 𝑒 𝑛𝑇𝑆 + 𝑞 𝑛𝑇𝑆 + 𝑥ො 𝑛𝑇𝑆
From eq(1)
= 𝑥 𝑛𝑇𝑆 + 𝑞 𝑛𝑇𝑆

• Input to the predictor is a quantized input sample and similarly 𝑏 𝑛𝑇𝑆 is


a binary coded output DPCM signal
• Predictor:
Prediction is a form of estimation with the help of the knowledge of past
samples. If X(n-1), X(n-2), … X(M-N) represents a random sample taken from a signal
෢𝑁 will be the predicted value, which is denoted as:
x(t), then 𝑋
𝑚
෢𝑁 = ෍ ℎ𝑂𝑘 𝑋𝑛−𝑘
𝑋
𝑘=1
෢𝑁 : The predicted value of the sample 𝑋𝑁
Where, 𝑋
m: No. of delay elements
ℎ𝑂𝑘 : Prediction co-efficient
𝑋𝑛−𝑘 :Past sample values

𝑋𝑛 𝑋𝑛−1 𝑋𝑛−2 𝑋𝑛−𝑚


Delay Delay Delay

ℎ𝑂1 X ℎ𝑂2 X ℎ𝑂𝑚 X

Ʃ Ʃ ෢𝑁
𝑋
DM [Delta Modulation]
• It is a 1-bit version of DPCM
• The input signal is oversampled i.e. 𝑓𝑠 ≫ 2𝑓𝑚 . Typically 𝑓𝑠 is set at
4 times the Nyquist rate, to increase the correlation between
adjacent samples.
• Advantages:
• Since each sample is encoded by 1-bit, there is no need of code word
formation.
• This system of waveform coding requires minimum channel bandwidth.
• The transmitter and the receiver of DM is much simpler.
• Delta Modulation produces a staircase approximation to the input
signal.
• The difference between the input signal x(t) and its approximated signal
u(t) is quantized into two levels of absolute value (±δ).
• If the difference in positive quantizer increases u(t) by +δ, the
difference with the negative quantizer reduces u(t) by –δ.
Primary encoded bits
transmitted DM

• A ‘0’ is transmitted when x(t) is greater than u(t) or the difference between
x(t) & u(t) is positive.
𝑒 𝑛𝑇𝑆 = 𝑥 𝑛𝑇𝑆 − 𝑥ො 𝑛𝑇𝑆
= 𝑥 𝑛𝑇𝑆 − 𝑢 𝑛𝑇𝑆 − 𝑇𝑆
𝑢 𝑛𝑇𝑆 = 𝑢 𝑛𝑇𝑆 − 𝑇𝑆 + 𝑏 𝑛𝑇𝑆
Where, 𝑏 𝑛𝑇𝑆 = 𝛿sgn[𝑒 𝑛𝑇𝑆 ]; output through quantizer
DM Transmitter
𝑥 𝑛𝑇𝑆 𝑒 𝑛𝑇𝑆 𝑏 𝑛𝑇𝑆
Ʃ 𝑄𝑢𝑎𝑛𝑡𝑖𝑧𝑒𝑟

𝑥ො 𝑛𝑇𝑆
Ʃ

𝐷𝑒𝑙𝑎𝑦 (𝑇𝑆 ) 𝑢 𝑛𝑇𝑆


𝑢 𝑛𝑇𝑆 − 𝑇𝑆
• DM Receiver:
Input is a binary signal 𝑏 𝑛𝑇𝑆 , the combination of delay 𝑇𝑆 and adder
produces the approximated signal u(t) from the DM signal. The low pass filter
cut-off of the input signal x(t) produces the output. The LPF is basically an
Integrator.

𝑏 𝑛𝑇𝑆 Approximated O/P


Ʃ LPF
Waveform x(t)

Delay
𝑇𝑆
• Drawbacks of DM System/ Noise effects in DM System/ Types of
Quantization error possible in DM System:
• Slope Overload Error: In order to collect the sequence of samples 𝑢 𝑛𝑇𝑆 from 𝑥 𝑛𝑇𝑆 ,
𝑢 𝑛𝑇𝑆 has to increase as fast as the input sequence/ signal x(t). Thus, the information
of the input message signal x(t) is lost till the DM encoded signal reaches the maximum
value.
Condition for no slope overload:
𝛿 𝑑𝑥(𝑡)
≫ 𝑚𝑎𝑥
𝑇𝑠 𝑑𝑡
If the condition for no slope overload is not fulfilled, then we find that the
approximated signal u(t) falls behind the input signal x(t), giving rise to a condition called
slope overload and the quantization error, thus developed is called slope overload
distortion.
• Granular Noise: If the stepsize δ is larger, relative to the local slope characteristics of the
input signal x(t), then the approximated signal u(t) hunts around the flat segment of the
input signal slope. This is known as Granular Noise.
ADM [Adaptive Delta Modulation]
• The stepsize ‘𝛿 ’ can be adaptive or changed according to the input signal
which was fixed in case of DM for which it can also be called as Linear
Delta Modulation (LDM).

• Transmitter:

Logic for
Step Size
Control

𝑥 𝑛𝑇𝑆 𝑒 𝑛𝑇𝑆 𝑏 𝑛𝑇𝑆


Ʃ 𝑄𝑢𝑎𝑛𝑡𝑖𝑧𝑒𝑟

𝑥ො 𝑛𝑇𝑆
Ʃ

𝐷𝑒𝑙𝑎𝑦 (𝑇𝑆 ) 𝑢 𝑛𝑇𝑆


𝑢 𝑛𝑇𝑆 − 𝑇𝑆
• Drawbacks of LDM can be minimized by using ADM, due to the introduction
of the ‘logic for stepsize control’. The algorithm for the stated logic is:

𝛿 𝑛𝑇𝑆 = 𝑔 𝑛𝑇𝑆 ∗ 𝛿 𝑛𝑇𝑆 − 𝑇𝑆


𝑘; 𝑏 𝑛𝑇𝑆 = 𝑏 𝑛𝑇𝑆 − 𝑇𝑆
where, 𝑔 𝑛𝑇𝑆 = ቊ
−𝑘; 𝑏 𝑛𝑇𝑆 ≠ 𝑏 𝑛𝑇𝑆 − 𝑇𝑆

• Receiver:

Approximated O/P
𝑏 𝑛𝑇𝑆 X Ʃ LPF
Waveform x(t)

Delay
Logic for
𝑇𝑆
Step Size
Control DM Receiver

Predictor
Digital Multiplexer

• It is used to multiplex digital signals having different bitrates.

• Digital signals, like digital data from computers, digitized voice


signal, digitized facsimile, etc. can be combined into a signal bit
string and transmitted over a single channel.

• Digital multiplexer uses the principle of bit by bit interleaving


procedure. A selector switch takes a narrow sample and time
division multiplexes the signal and transmits them onto a single
channel or a common line.
There are two types of digital multiplexers:

Low Speed MUX: This is designed to combine low speed digital


signals upto a max rate of 4800bps to a high speed of max 9600bps.
These low speed MUX are used to transmit data over voice channels
of telephone network. ex.: The working of a MODEM.

High Speed MUX: This is designed to operate at much higher bitrate


and is used in data transmission. The digital error (e) is based on (T1)
carrier is high speed MUX. It requires a number of MUX to design and
operate at different bitrates.
• The revolution of the commutator key is 8000bps.
• This system accommodates 24 voice channels transmitted at a bit rate
1.544Mbps.
* The sampling rate of telephone signal is 8kHz.
Bit Frame
• The commutator sweeps continuously from S1 to S24 and
then back to S1 at a rate of 8kHz/8000bps. This provides
8000 samples per second for each signal.
• The sampling rate of telephone signal is 8kHz, this process
interleaves the samples and produces time division
multiplexed signals.
• Each signal is then encoded in 8bits (n=8) corresponding
to 256 quantization levels.
• The digital signal thus generated during 1 complete
revolution of the commutator contains:
24 x 8 = 192bits
• Each encoded 8bits, L=256.
• Thus, for 24 channels, the commutator produces digital
signal of 192bits.
• It is again added with 1 interleaving bit to makeup the bit
frame. i.e. 192+1=193bits/frame.
Frame Synchronization
• The extra bit added is called the synchronization bit or the
interleaving bit, is used at the receiver as a string of bits
that correspond to the original signal.

Bit Rate
• Each signal is sampled 8000 times per second so that a
complete frame of period:
1
𝑇𝑃 = = 0.125 × 10−3 𝑠 = 125𝜇𝑠
8000
• This 𝑇𝑃 accommodates 193bits. Thus,

𝑁𝑜. 𝑜𝑓 𝑏𝑖𝑡𝑠 193 6 𝑏𝑝𝑠


𝐵𝑖𝑡 𝑅𝑎𝑡𝑒 = = = 1.54 × 10
𝑇𝑖𝑚𝑒 𝑃𝑒𝑟𝑖𝑜𝑑 0.125 × 10−6

=1.54Mbps
Hierarchy Diagram/ T1 Carrier:

T1 T1 T1
S1
S2 T2 T2 T2
PCM T3 M12 M23 M34
S24 T4 T7 T6

Bits per frame of M12: Bits per frame of M23:


(193 x 4) + 17 = 789bits/frame (789 x 7) + 69 = 5592bits/frame
Bit Rate = bits/frame x frames/second Bit Rate = bits/frame x frames/second
= 789 x 8000 = 5592 x 8000
= 6.312Mbps = 44.736Mbps
Here, 17 interleaving bits are added. Here, 69 interleaving bits are added.

Bits per frame of M34:


(5542 x 6) + 720 = 34272bits/frame
Bit Rate = bits/frame x frames/second
= 34272 x 8000
= 274.17Mbps
Here, 720 interleaving bits are added.

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