DC Unit-2 Baseband Transmission

2.1 Line Coding and its properties

A line code is the code used for data transmission of a digital signal over a transmission
line. This process of coding is chosen so as to avoid overlap and distortion of signal
such as inter-symbol interference.
Properties of Line Coding
Following are the properties of line coding −
 As the coding is done to make more bits transmit on a single signal, the bandwidth
used is much reduced.
 For a given bandwidth, the power is efficiently used.
 The probability of error is much reduced.
 Error detection is done and the bipolar too has a correction capability.
 Power density is much favourable.
 The timing content is adequate.
 Long strings of 1s and 0s is avoided to maintain transparency.
Types of Line Coding
There are 3 types of Line Coding
 Unipolar
 Polar
 Bi-polar
Unipolar Signaling
Unipolar signaling is also called as On-Off Keying or simply OOK.
The presence of pulse represents a 1 and the absence of pulse represents a 0.
There are two variations in Unipolar signaling −
 Non-Return to Zero NRZ
 Return to Zero RZRZ
Unipolar Non-Return to Zero NRZ
In this type of unipolar signaling, a High in data is represented by a positive pulse called
as Mark, which has a duration T0 equal to the symbol bit duration. A Low in data input
has no pulse.
The following figure clearly depicts this.

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 Fig 1 Uni-polar NRZ
Advantages
The advantages of Unipolar NRZ are −
 It is simple.
 A lesser bandwidth is required.
Disadvantages
The disadvantages of Unipolar NRZ are −
 No error correction done.
 Presence of low frequency components may cause the signal droop.
 No clock is present.
 Loss of synchronization is likely to occur (especially for long strings of 1s and 0s).
Unipolar Return to Zero RZ

In this type of unipolar signaling, a High in data, though represented by a Mark pulse,
its duration T0 is less than the symbol bit duration. Half of the bit duration remains high
but it immediately returns to zero and shows the absence of pulse during the remaining
half of the bit duration.
It is clearly understood with the help of the following figure.

Fig 2 Unipolar RZ
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 Advantages
The advantages of Unipolar RZ are −
 It is simple.
 The spectral line present at the symbol rate can be used as a clock.
Disadvantages
The disadvantages of Unipolar RZ are −
 No error correction.
 Occupies twice the bandwidth as unipolar NRZ.
 The signal droop is caused at the places where signal is non-zero at 0 Hz.
Polar Signaling
There are two methods of Polar Signaling. They are −
 Polar NRZ
 Polar RZ
Polar NRZ
In this type of Polar signaling, a High in data is represented by a positive pulse, while
a Low in data is represented by a negative pulse. The following figure depicts this well.

Fig 3 Polar NRZ

Advantages
The advantages of Polar NRZ are −
 It is simple.
 No low-frequency components are present.

Disadvantages
The disadvantages of Polar NRZ are −
 No error correction.
 No clock is present.
 The signal droop is caused at the places where the signal is non-zero at 0 Hz.
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Polar RZ
In this type of Polar signaling, a High in data, though represented by a Mark pulse, its
duration T0 is less than the symbol bit duration. Half of the bit duration remains high but
it immediately returns to zero and shows the absence of pulse during the remaining half
of the bit duration.
However, for a Low input, a negative pulse represents the data, and the zero level
remains same for the other half of the bit duration. The following figure depicts this
clearly.

Fig 4 Polar RZ
Advantages
The advantages of Polar RZ are −
 It is simple.
 No low-frequency components are present.

Disadvantages
The disadvantages of Polar RZ are −
 No error correction.
 No clock is present.
 Occupies twice the bandwidth of Polar NRZ.
 The signal droop is caused at places where the signal is non-zero at 0 Hz.

Bipolar Signaling
This is an encoding technique which has three voltage levels namely +, - and 0. Such a
signal is called as duo-binary signal.
An example of this type is Alternate Mark Inversion AMIAMI. For a 1, the voltage level
gets a transition from + to – or from – to +, having alternate 1s to be of equal polarity.
A 0 will have a zero-voltage level.
Even in this method, we have two types.
 Bipolar NRZ
 Bipolar RZ
From the models so far discussed, we have learnt the difference between NRZ and RZ.
It just goes in the same way here too. The following figure clearly depicts this.
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 Fig 5 Bipolar RZ and Bipolar NRZ

The above figure has both the Bipolar NRZ and RZ waveforms. The pulse duration and
symbol bit duration are equal in NRZ type, while the pulse duration is half of the symbol
bit duration in RZ type.
Advantages

Following are the advantages −

 It is simple.
 No low-frequency components are present.
 Occupies low bandwidth than unipolar and polar NRZ schemes.
 This technique is suitable for transmission over AC coupled lines, as
signal drooping doesn’t occur here.
 A single error detection capability is present in this.
Disadvantages
Following are the disadvantages −
 No clock is present.
 Long strings of data causes loss of synchronization.
Power Spectral Density

The function which describes how the power of a signal got distributed at various
frequencies, in the frequency domain is called as Power Spectral Density PSD.
PSD is the Fourier Transform of Auto-Correlation Similarity between observations. It is
in the form of a rectangular pulse.

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 Fig 6 PSD
Key Takeaways:

 Line coding determines how the binary data is represented on the link.
 The mark density (MD) of a binary data pattern is defined as the number of one bit in
the pattern, divided by the length of the pattern:

2.2 Various types of PCM waveforms

Pulse code modulation is a method that is used to convert an analog signal into a digital
signal so that a modified analog signal can be transmitted through the digital
communication network. PCM is in binary form, so there will be only two possible states
high and low (0 and 1). We can also get back our analog signal by demodulation. The
Pulse Code Modulation process is done in three steps Sampling, Quantization, and
Coding. There are two specific types of pulse code modulations such as differential
pulse code modulation (DPCM) and adaptive differential pulse code modulation
(ADPCM)
2.3 Attributes of PCM waveforms
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.

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Basic Elements of PCM
The transmitter section comprises of Sampling, Quantizing and Encoding. 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.

Fig.7: PCM
Low Pass Filter
This filter eliminates the high frequency components present in the input analog signal
to avoid aliasing of the message signal.
Sampler
It helps to collect the sample data at instantaneous values of message signal, so as to
reconstruct the original signal. The sampling rate must be in accordance with the
sampling theorem.
Quantizer
It reduces excessive bits and confines the data. It reduces the redundant bits and
compresses the value of the sampled output.
Encoder
The digitization of analog signal is done by the encoder. It designates each quantized
level by a binary code. Encoding minimizes the bandwidth used.
Regenerative Repeater
It 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.

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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.
Key takeaway
The transmitter section comprises of Sampling, Quantizing and Encoding. 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.
2.4 Mary Pulse Modulation waveforms
This is called as M-ary Phase Shift Keying M−ary PSK
The phase of the carrier signal, takes on M different levels.
Representation of M-ary PSK

Si(t)=√2E/T cos(wot +ϕit) 0≤t≤T and i=1,2...M

ϕi(t)=2πiM where i=1,2,3......M

Some prominent features of M-ary PSK are −

 The envelope is constant with more phase possibilities.
 This method was used during the early days of space communication.
 Better performance than ASK and FSK.
 Minimal phase estimation error at the receiver.
 The bandwidth efficiency of M-ary PSK decreases and the power efficiency
increases with the increase in M.
So far, we have discussed different modulation techniques. The output of all these
techniques is a binary sequence, represented as 1s and 0s
Multi-level modulation techniques permit high data rates within fixed
bandwidth constraints. A convenient set of signals for M-ary PSK is

Where the M phase angles are

For equiprobable ones and zeros the PSD for M-ary PSK is

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The symbols in this case are of duration , so the information (or bit) rate satisfies

The potential bandwidth efficiency of M-ary PSK can be shown to be

A phase diagram and signal constellation diagram for the case of M = 8 are shown
below:

Fig 8 Phasor and Signal Constellation Diagram

The probability of error calculation involves analysing the received phase at the
receiver, (in the presence of noise), and comparing it to the actual phases. An exact
solution is difficult to compute, but for an approximate probability of making a
symbol error is

If a Gray code is used, then the corresponding bit error is approximately

M-ary QAM
The QAM modulator essentially follows the idea that can be seen from the basic QAM
theory where there are two carrier signals with a phase shift of 90° between them. These
are then amplitude modulated with the two data streams known as the I or In-phase and
the Q or quadrature data streams. These are generated in the baseband processing
area.
Basic QAM I-Q modulator circuit

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The two resultant signals are summed and then processed as required in the RF signal
chain, typically converting them in frequency to the required final frequency and
amplifying them as required.

Fig 9 QAM Modulator

It is worth noting that as the amplitude of the signal varies any RF amplifiers must be
linear to preserve the integrity of the signal. Any non-linearities will alter the relative
levels of the signals and alter the phase difference, thereby distorting he signal and
introducing the possibility of data errors.
Basic QAM I-Q demodulator circuit
The basic modulator assumes that the two quadrature signals remain exactly in
quadrature.
A further requirement is to derive a local oscillator signal for the demodulation that is
exactly on the required frequency for the signal. Any frequency offset will be a change
in the phase of the local oscillator signal with respect to the two double sideband
suppressed carrier constituents of the overall signal.
Systems include circuitry for carrier recovery that often utilises a phase locked loop -
some even have an inner and outer loop. Recovering the phase of the carrier is
important otherwise the bit error rate for the data will be compromised.

Fig 10 QAM Demodulator

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The circuits shown above show the generic IQ QAM modulator and demodulator
circuits that are used in a vast number of different areas. Not only are these circuits
made from discrete components, but more commonly they are used within integrated
circuits that are able to provide a large number of functions.
Error Probability

Key takeaway

The QAM modulator essentially follows the idea that can be seen from the basic QAM
theory where there are two carrier signals with a phase shift of 90° between them.
These are then amplitude modulated with the two data streams known as the I or In-
phase and the Q or quadrature data streams.
The phase of the carrier signal, takes on M different levels.
Representation of M-ary PSK

Si(t)=√2E/T cos(wot +ϕit) 0≤t≤T and i=1,2...M

ϕi(t)=2πiM where i=1,2,3......M

2.5 Differential Pulse Code Modulation

Differential pulse code modulation (DPCM) is a procedure of converting an analog
into a digital signal in which an analog signal is sampled and then the difference
between the actual sample value and its predicted value (predicted value is based on
previous sample or samples) is quantized and then encoded forming a digital value.
DPCM code words represent differences between samples unlike PCM where code
words represented a sample value.
Basic concept of DPCM - coding a difference, is based on the fact that most source
signals show significant correlation between successive samples so encoding uses
redundancy in sample values which implies lower bit rate.
Realization of basic concept (described above) is based on a technique in which we
have to predict current sample value based upon previous samples (or sample) and
we have to encode the difference between actual value of sample and predicted value
(the difference between samples can be interpreted as prediction error).
Because it's necessary to predict sample value DPCM is form of predictive coding.
DPCM compression depends on the prediction technique, well-conducted prediction
techniques lead to good compression rates, in other cases DPCM could mean
expansion comparing to regular PCM encoding.

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 Fig.11: DPCM
Key Takeaways:
 DPCM code words represent differences between samples unlike PCM where code
words represented a sample value.
 Basic concept of DPCM - coding a difference, is based on the fact that most source
signals show significant correlation between successive samples so encoding uses
redundancy in sample values which implies lower bit rate.
2.6 Multiplexing of PCM signals
In pulse modulation the time division multiplexing makes the maximum utilisation of the
transmission channel. The TDM can be analog or digital. The digital multiplexing is
discussed below. The block diagram for TDM is shown below. N PAM channels are
multiplexed and each channel to be transmitted is passed through the LPF. The input to
the commutators is the signal coming out from the LPF. The commutator rotates at the
rate of fs. A t the receiver the demodulator separates the time multiplexed input
channels. These signal pass through the low-pass reconstruction filters. If W is the
highest signal frequency then according to the sampling theorem fs 2W

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 Fig 12 TDM system block diagram and waveform [Ref 6]
The time space between two samples is given by
Ts = 1/fs
Ts 2W
If there are N input channels then the space between two samples is given by
Space between two samples = Ts/N
The number of pulses/secs are called as signalling rate of TDM denoted by r
r= Nfs
For PAM/TDM Signalling rate r 2NW
Synchronisation in TDM
There should always be synchronisation in transmitter and receiver. Usually,
markers are inserted to indicate the separation between the frames. Synchronisation
is obtained due to these marker pulses but the number of channels to be multiplexed
is reduced by one. They become N-1.
Crosstalk
In TDM transmission the signal is converted to smooth modulating waveform when
passed through the baseband filter. The baseband value passes through the values
of all the individual samples.

Fig 13 TDM transmission with baseband filtering and its waveform

Due to this baseband filter crosstalk arises in between two samples. This interference
can be reduced by increasing the distance between individual signal samples. The
minimum distance required to avoid crosstalk between two samples is called as
guard time.

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 Fig 14 Applied rectangular pulse and its response
From above figure we see that even after the pulse is removed the response decays
from its value A and then takes longer period. The guard time Tg represents
minimum pulse spacing. After guard time end the pulse tail is less than Act.
Act = A
B= Bandwidth of signal
Key takeaway
The time space between two samples is given by
Ts = 1/fs
Ts 2W
The number of pulses/secs are called as signalling rate of TDM denoted by r
r= Nfs
For PAM/TDM Signalling rate r 2NW

Example
Twelve different message signals each of bandwidth 20kHz are to be
multiplexed and transmitted. Determine the minimum bandwidth required for
PAM/TDM system?
Sol: No. of channels N =12
Bandwidth of each channel fm = 20kHz
Minimum bandwidth required to avoid crosstalk is = Nfm = 20 x 12 = 240kHz

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2.7 Delta modulation
The type of modulation, where the sampling rate is much higher and in which the
step size after quantization is of a smaller value Δ, such a modulation is termed
as delta modulation.
Features
 The quality is moderate.
 The design of the modulator and the demodulator is simple.
 The stair-case approximation of output waveform.
 The step-size is very small, i.e., Δ delta.
 The bit rate can be decided by the user.
 This involves simpler implementation.
Delta Modulation is a simplified form of DPCM technique, also viewed as 1-bit
DPCM scheme. As the sampling interval is reduced, the signal correlation will be
higher.
Delta Modulator
The Delta Modulator comprises of a 1-bit quantizer and a delay circuit along with
two summer circuits. Following is the block diagram of a delta modulator.

Fig.15: Delta Modulator

The predictor circuit in DPCM is replaced by a simple delay circuit in DM.
From the above diagram, we have the notations as −
 x(nTs)= over sampled input
 ep(nTs) = summer output and quantizer input
 eq(nTs) = quantizer output = v(nTs)
 xˆ(nTs) = output of delay circuit
 u(nTs) = input of delay circuit
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Using these notations, now we shall try to figure out the process of delta modulation.
ep(nTs)=x(nTs)−xˆ(nTs) ---------equation 1
=x(nTs)−u([n−1]Ts
=x(nTs)−[xˆ[[n−1]Ts]+v[[n−1]Ts]] -------equation 2
Further,
v(nTs)=eq(nTs)=S.sig.[ep(nTs)] ---------equation 3
u(nTs)=xˆ(nTs)+eq(nTs)
Where,
 xˆ(nTs) = the previous value of the delay circuit
 eq(nTs) = quantizer output = v(nTs)
Hence,
u(nTs)=u([n−1]Ts)+v(nTs) ---------equation 4
Which means,
The present input of the delay unit
= The previous output of the delay unit + the present quantizer output the present
quantizer output
Assuming zero condition of Accumulation,

Accumulated version of DM output = --------equation 5

Now, note that
xˆ(nTs)=u([n−1]Ts)

= ---------equation 6
Delay unit output is an Accumulator output lagging by one sample.
From equations 5 & 6, we get a possible structure for the demodulator.
A Stair-case approximated waveform will be the output of the delta modulator with
the step-size as delta (Δ). The output quality of the waveform is moderate.

Delta Demodulator
The delta demodulator comprises of a low pass filter, a summer, and a delay circuit.
The predictor circuit is eliminated here and hence no assumed input is given to the
demodulator.
Following is the diagram for delta demodulator.

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 Fig.16 Delta Demodulator
From the above diagram, we have the notations as −
 vˆ(nTs) is the input sample
 uˆ(nTs) is the summer output
 x¯(nTs) is the delayed output
A binary sequence will be given as an input to the demodulator. The stair-case
approximated output is given to the LPF.
Low pass filter is used for many reasons, but the prominent reason is noise
elimination for out-of-band signals. The step-size error that may occur at the
transmitter is called granular noise, which is eliminated here. If there is no noise
present, then the modulator output equals the demodulator input.
Advantages of DM Over DPCM
 1-bit quantizer
 Very easy design of the modulator and the demodulator
Key takeaway
The type of modulation, where the sampling rate is much higher and in which the
step size after quantization is of a smaller value Δ, such a modulation is termed
as delta modulation.
2.8 Idling noise and slope overload
Delta modulation is subject to rate of rise over load problems whenever the input
changes too rapidly for the stepped wave form to follow it. If the input signal level
remains constant, the reconstructed Delta modulation waveform exhibits a hunting
behavior known as idling noise. This idling noise is a square wave at one half the
clock rate. If the clock rate is much greater than twice the highest frequency in the
input signal, most of the idling noise can be filtered out at the receiver.

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 Fig 17 Slope Overload
Slope Overload: Slope overload distortion occurs when the analog input signal
changes at a faster rate than the DAC can maintain it, the slope of the analog signal is
greater than the delta modulator can maintain. In general, when the slope of stair
case is less than (or) equal to modulating signal, the slope overloading occurs.
Increasing the clock frequency reduces the probability of slope overload occurring.
General method to reduce the slope overload is to increase the magnitude of the size.
Assume the input for Delta modulation be f(t) = A cos ωmt.

If the step size used in the Delta modulation system is ‘Δ’, then the maximum (rate of rise) slope
over load is / Ts
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Granular noise: In general, Granular noise occurs
 When the original analog input signal has a relatively constant amplitude, the
reconstructed signal has variations that were not present in the original signal. This
is called granular noise
 The granular noise is analogous to the quantization noise in a PCM system.
 When the step size Δ is too large relative to the local slope characteristics of the
input waveform x(t), thereby causing the stair case approximation u(t) to hunt
around a relatively flat segment of the input waveform Granular noise can be
reduced by decreasing the step size so that the stair case approximation may
become more closer to the modulating signal.
There is a need to have a large step size to accommodate a wide dynamic range,
whereas small step size is required for the accurate representation of relatively low-
level signals.

Therefore, a granular noise can be removed by taking a small resolution of step size
and a slope over load distortion can be removed by taking a large resolution.
Key takeaway
The input for Delta modulation be f(t) = A cos ωmt.

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2.9 Adaptive Delta Modulation
In digital modulation, we have come across certain problem of determining the step-
size, which influences the quality of the output wave.
A larger step-size is needed in the steep slope of modulating signal and a smaller
step size is needed where the message has a small slope. The minute details get
missed in the process. So, it would be better if we can control the adjustment of step-
size, according to our requirement in order to obtain the sampling in a desired
fashion. This is the concept of Adaptive Delta Modulation.
Following is the block diagram of Adaptive delta modulator.

Fig. 18 Adaptive delta modulator

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The gain of the voltage-controlled amplifier is adjusted by the output signal from the sampler. The
amplifier gain determines the step-size and both are proportional.
ADM quantizes the difference between the value of the current sample and the predicted value of
the next sample. It uses a variable step height to predict the next values, for the faithful
reproduction of the fast-varying values.
Key takeaway
ADM quantizes the difference between the value of the current sample and the predicted value of
the next sample. It uses a variable step height to predict the next values
2.10 Adaptive DPCM
Adaptive Differential Pulse Code Modulation (ADPCM), also called Delta Pulse Code Modulation,
is a compression coding for a signal, which is based on differential values similar to Differential
Pulse Code Modulation (DPCM), and in which the scaling of the quantization stages is additionally
adjusted (adapted) depending on the signal curve.
ADPCM is a pulse code modulation with prediction function. When the signal is processed, an
attempt is made to predict the further course of the signal within the next section. For the
quantization of the signal in the next time step only the difference between predicted and real
signal is used. Due to this difference formation, fewer bits can be used to describe the signal.
With this method, both the prediction function and the quantization level are "adapted" anew with
each work step. This control loop provides a better signal prediction than DPCM.
2.11 Comparison of PCM and DM

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