Chapter 2: Physical Layer

2.2 Digital Transmission

4.1 Copyright © The McGraw-Hill Companies, Inc. Permission required for reproduction or display.
 2.2-1 DIGITAL-TO-DIGITAL CONVERSION

In this section, we see how we can represent digital

data by using digital signals. The conversion involves
three techniques: line coding, block coding, and
scrambling. Line coding is always needed; block
coding and scrambling may or may not be needed.

Topics discussed in this section:

Line Coding
Line Coding Schemes
Block Coding
Scrambling

4.2
 Figure 2.2.1 Line coding and decoding

4.3
 Mapping Data symbols onto
Signal levels
 A data symbol (or element) can consist of a
number of data bits:
 1 , 0 or
 11, 10, 01, ……
 A data symbol can be coded into a single
signal element or multiple signal elements
 1 -> +V, 0 -> -V
 1 -> +V and -V, 0 -> -V and +V
 The ratio „r‟ is the number of data elements
carried by a signal element.
4.4
 Relationship between data
rate and signal rate
 The data rate defines the number of bits sent
per sec - bps. It is often referred to the bit
rate.
 The signal rate is the number of signal
elements sent in a second and is measured in
bauds. It is also referred to as the modulation
rate.
 Goal is to increase the data rate whilst
reducing the baud rate.

4.5
 Figure 2.2.2 Signal element versus data element

4.6
 Data rate and Baud rate

 The baud or signal rate can be

expressed as:
S = c x N x 1/r bauds
where N is data rate
c is the case factor (worst, best & avg.)
r is the ratio between data element &
signal element

4.7
 Example 2.2.1

A signal is carrying data in which one data element is

encoded as one signal element ( r = 1). If the bit rate is
100 kbps, what is the average value of the baud rate if c is
between 0 and 1?

Solution
We assume that the average value of c is 1/2 . The baud
rate is then

4.8
 Note

Although the actual bandwidth of a

digital signal is infinite, the effective
bandwidth is finite.

4.9
 Example 2.2.2

The maximum data rate of a channel (see Chapter 3) is

Nmax = 2 × B × log2 L (defined by the Nyquist formula).
Does this agree with the previous formula for Nmax?

Solution
A signal with L levels actually can carry log2L bits per
level. If each level corresponds to one signal element and
we assume the average case (c = 1/2), then we have

4.10
 Considerations for choosing a good
signal element referred to as line
encoding
 Baseline wandering - a receiver will evaluate
the average power of the received signal
(called the baseline) and use that to determine
the value of the incoming data elements. If
the incoming signal does not vary over a long
period of time, the baseline will drift and thus
cause errors in detection of incoming data
elements.
 A good line encoding scheme will prevent long
runs of fixed amplitude.
4.11
 Line encoding C/Cs

 DC components - when the voltage

level remains constant for long periods
of time, there is an increase in the low
frequencies of the signal. Most channels
are bandpass and may not support the
low frequencies.
 This will require the removal of the dc
component of a transmitted signal.
4.12
 Line encoding C/Cs

 Self synchronization - the clocks at the

sender and the receiver must have the
same bit interval.
 If the receiver clock is faster or slower it
will misinterpret the incoming bit
stream.

4.13
 Figure 2.2.3 Effect of lack of synchronization

4.14
 Example 2.2.3

In a digital transmission, the receiver clock is 0.1 percent

faster than the sender clock. How many extra bits per
second does the receiver receive if the data rate is
1 kbps? How many if the data rate is 1 Mbps?
Solution
At 1 kbps, the receiver receives 1001 bps instead of 1000
bps.

At 1 Mbps, the receiver receives 1,001,000 bps instead of

1,000,000 bps.

4.15
 Line encoding C/Cs

 Error detection - errors occur during

transmission due to line impairments.
 Some codes are constructed such that
when an error occurs it can be
detected. For example: a particular
signal transition is not part of the code.
When it occurs, the receiver will know
that a symbol error has occurred.
4.16
 Line encoding C/Cs

 Noise and interference - there are line

encoding techniques that make the
transmitted signal “immune” to noise
and interference.
 This means that the signal cannot be
corrupted, it is stronger than error
detection.

4.17
 Line encoding C/Cs

 Complexity - the more robust and

resilient the code, the more complex it
is to implement and the price is often
paid in baud rate or required
bandwidth.

4.18
 Figure 2.2.4 Line coding schemes

4.19
 Figure 2.2.5 Unipolar NRZ scheme

4.20
 Figure 2.2.6 Polar NRZ-L and NRZ-I schemes

4.21
 Note

In NRZ-L the level of the voltage

determines the value of the bit.
In NRZ-I the inversion
or the lack of inversion
determines the value of the bit.

4.22
 Note

NRZ-L and NRZ-I both have an average

signal rate of N/2 Bd.

4.23
 Note

NRZ-L and NRZ-I both have a DC

component problem.

4.24
 Example 2.2.4

A system is using NRZ-I to transfer 10-Mbps data. What

are the average signal rate and minimum bandwidth?

Solution
The average signal rate is S = N/2 = 500 kbaud. The
minimum bandwidth for this average baud rate is Bmin =
S = 500 kHz.

4.25
 Figure 2.2.7 Polar RZ scheme

4.26
 Figure 2.2.8 Polar biphase: Manchester and differential Manchester schemes

4.27
 Note

In Manchester and differential

Manchester encoding, the transition
at the middle of the bit is used for
synchronization.

4.28
 Note

The minimum bandwidth of Manchester

and differential Manchester is 2 times
that of NRZ.

4.29
 Note

In bipolar encoding, we use three levels:

positive, zero, and negative.

4.30
 Figure 2.2.9 Bipolar schemes: AMI and pseudoternary

4.31
 Note

In mBnL schemes, a pattern of m data

elements is encoded as a pattern of n
signal elements in which 2m ≤ Ln.

4.32
 Figure 2.2.10 Multilevel: 2B1Q scheme

4.33
 Figure 2.2.11 Multilevel: 8B6T scheme

4.34
 Figure 2.2.12 Multilevel: 4D-PAM5 scheme

4.35
 Figure 2.2.13 Multitransition: MLT-3 scheme

4.36
 Table 2.2.1 Summary of line coding schemes

4.37
 Note

Block coding is normally referred to as

mB/nB coding;
it replaces each m-bit group with an
n-bit group.

4.38
 Figure 2.2.14 Block coding concept

4.39
 Figure 2.2.15 Using block coding 4B/5B with NRZ-I line coding scheme

4.40
Table 2.2.2 4B/5B mapping codes

4.41
 Figure 2.2.16 Substitution in 4B/5B block coding

4.42
 Example 2.2.5

We need to send data at a 1-Mbps rate. What is the

minimum required bandwidth, using a combination of
4B/5B and NRZ-I or Manchester coding?

Solution
First 4B/5B block coding increases the bit rate to 1.25
Mbps. The minimum bandwidth using NRZ-I is N/2 or
625 kHz. The Manchester scheme needs a minimum
bandwidth of 1 MHz. The first choice needs a lower
bandwidth, but has a DC component problem; the second
choice needs a higher bandwidth, but does not have a DC
component problem.
4.43
 Figure 2.2.17 8B/10B block encoding

4.44
 Figure 2.2.18 AMI used with scrambling

4.45
 Figure 2.2.19 Two cases of B8ZS scrambling technique

4.46
 Note

B8ZS substitutes eight consecutive

zeros with 000VB0VB.

4.47
 Figure 2.2.20 Different situations in HDB3 scrambling technique

4.48
 Note

HDB3 substitutes four consecutive

zeros with 000V or B00V depending
on the number of nonzero pulses after
the last substitution.

4.49
 2.2-2 ANALOG-TO-DIGITAL CONVERSION

A digital signal is superior to an analog signal because

it is more robust to noise and can easily be recovered,
corrected and amplified. For this reason, the tendency
today is to change an analog signal to digital data. In
this section we describe two techniques, pulse code
modulation and delta modulation.

Topics discussed in this section:

 Pulse Code Modulation (PCM)
 Delta Modulation (DM)

4.50
 PCM
 PCM consists of three steps to digitize an
analog signal:
1. Sampling
2. Quantization
3. Binary encoding
 Before we sample, we have to filter the
signal to limit the maximum frequency of
the signal as it affects the sampling rate.
 Filtering should ensure that we do not
distort the signal, ie remove high frequency
components that affect the signal shape.

4.51
 Figure 2.2.21 Components of PCM encoder

4.52
 Sampling
 Analog signal is sampled every TS secs.
 Ts is referred to as the sampling interval.
 fs = 1/Ts is called the sampling rate or
sampling frequency.
 There are 3 sampling methods:
 Ideal - an impulse at each sampling instant
 Natural - a pulse of short width with varying
amplitude
 Flattop - sample and hold, like natural but with
single amplitude value
 The process is referred to as pulse amplitude
modulation PAM and the outcome is a signal
with analog (non integer) values
4.53
 Figure 2.2.22 Three different sampling methods for PCM

4.54
 Note

According to the Nyquist theorem, the

sampling rate must be
at least 2 times the highest frequency
contained in the signal.

4.55
 Figure 2.2.23 Nyquist sampling rate for low-pass and bandpass signals

4.56
 Example 2.2.6

For an intuitive example of the Nyquist theorem, let us

sample a simple sine wave at three sampling rates: fs = 4f
(2 times the Nyquist rate), fs = 2f (Nyquist rate), and
fs = f (one-half the Nyquist rate). Figure 4.24 shows the
sampling and the subsequent recovery of the signal.

It can be seen that sampling at the Nyquist rate can create

a good approximation of the original sine wave (part a).
Oversampling in part b can also create the same
approximation, but it is redundant and unnecessary.
Sampling below the Nyquist rate (part c) does not produce
a signal that looks like the original sine wave.
4.57
 Figure 2.2.24 Recovery of a sampled sine wave for different sampling rates

4.58
 Example 2.2.7

Consider the revolution of a hand of a clock. The second

hand of a clock has a period of 60 s. According to the
Nyquist theorem, we need to sample the hand every 30 s
(Ts = T or fs = 2f ). In Figure 4.25a, the sample points, in
order, are 12, 6, 12, 6, 12, and 6. The receiver of the
samples cannot tell if the clock is moving forward or
backward. In part b, we sample at double the Nyquist rate
(every 15 s). The sample points are 12, 3, 6, 9, and 12.
The clock is moving forward. In part c, we sample below
the Nyquist rate (Ts = T or fs = f ). The sample points are
12, 9, 6, 3, and 12. Although the clock is moving forward,
the receiver thinks that the clock is moving backward.
4.59
 Figure 2.2.25 Sampling of a clock with only one hand

4.60
 Example 2.2.8

An example related to Example 4.7 is the seemingly

backward rotation of the wheels of a forward-moving car
in a movie. This can be explained by under-sampling. A
movie is filmed at 24 frames per second. If a wheel is
rotating more than 12 times per second, the under-
sampling creates the impression of a backward rotation.

4.61
 Example 2.2.9

Telephone companies digitize voice by assuming a

maximum frequency of 4000 Hz. The sampling rate
therefore is 8000 samples per second.

4.62
 Example 2.2.10

A complex low-pass signal has a bandwidth of 200 kHz.

What is the minimum sampling rate for this signal?

Solution
The bandwidth of a low-pass signal is between 0 and f,
where f is the maximum frequency in the signal.
Therefore, we can sample this signal at 2 times the
highest frequency (200 kHz). The sampling rate is
therefore 400,000 samples per second.

4.63
 Example 2.2.11

A complex bandpass signal has a bandwidth of 200 kHz.

What is the minimum sampling rate for this signal?

Solution
We cannot find the minimum sampling rate in this case
because we do not know where the bandwidth starts or
ends. We do not know the maximum frequency in the
signal.

4.64
 Quantization
 Sampling results in a series of pulses of
varying amplitude values ranging between
two limits: a min and a max.
 The amplitude values are infinite between the
two limits.
 We need to map the infinite amplitude values
onto a finite set of known values.
 This is achieved by dividing the distance
between min and max into L zones, each of
height 
 = (max - min)/L
4.65
 Quantization Levels

 The midpoint of each zone is assigned a

value from 0 to L-1 (resulting in L
values)
 Each sample falling in a zone is then
approximated to the value of the
midpoint.

4.66
 Quantization Zones
 Assume we have a voltage signal with
amplitutes Vmin=-20V and Vmax=+20V.
 We want to use L=8 quantization levels.
 Zone width  = (20 - -20)/8 = 5
 The 8 zones are: -20 to -15, -15 to -10,
-10 to -5, -5 to 0, 0 to +5, +5 to +10,
+10 to +15, +15 to +20
 The midpoints are: -17.5, -12.5, -7.5, -
2.5, 2.5, 7.5, 12.5, 17.5

4.67
 Assigning Codes to Zones
 Each zone is then assigned a binary code.
 The number of bits required to encode the
zones, or the number of bits per sample as it
is commonly referred to, is obtained as
follows:
nb = log2 L
 Given our example, nb = 3
 The 8 zone (or level) codes are therefore:
000, 001, 010, 011, 100, 101, 110, and 111
 Assigning codes to zones:
 000 will refer to zone -20 to -15
 001 to zone -15 to -10, etc.
4.68
 Figure 4.26 Quantization and encoding of a sampled signal

4.69
 Quantization Error
 When a signal is quantized, we introduce an
error - the coded signal is an approximation
of the actual amplitude value.
 The difference between actual and coded
value (midpoint) is referred to as the
quantization error.
 The more zones, the smaller  which results
in smaller errors.
 BUT, the more zones the more bits required
to encode the samples -> higher bit rate

4.70
 Quantization Error and SNQR
 Signals with lower amplitude values will suffer
more from quantization error as the error
range: /2, is fixed for all signal levels.
 Non linear quantization is used to alleviate
this problem. Goal is to keep SNQR fixed for
all sample values.
 Two approaches:
 The quantization levels follow a logarithmic curve.
Smaller ‟s at lower amplitudes and larger ‟s at
higher amplitudes.
 Companding: The sample values are compressed
at the sender into logarithmic zones, and then
expanded at the receiver. The zones are fixed in
height.
4.71
 Bit rate and bandwidth
requirements of PCM
 The bit rate of a PCM signal can be calculated form
the number of bits per sample x the sampling rate
Bit rate = nb x fs
 The bandwidth required to transmit this signal
depends on the type of line encoding used. Refer to
previous section for discussion and formulas.
 A digitized signal will always need more bandwidth
than the original analog signal. Price we pay for
robustness and other features of digital transmission.

4.72
 Example 4.14

We want to digitize the human voice. What is the bit rate,

assuming 8 bits per sample?

Solution
The human voice normally contains frequencies from 0
to 4000 Hz. So the sampling rate and bit rate are
calculated as follows:

4.73
 PCM Decoder

 To recover an analog signal from a digitized

signal we follow the following steps:
 We use a hold circuit that holds the amplitude
value of a pulse till the next pulse arrives.
 We pass this signal through a low pass filter with
a cutoff frequency that is equal to the highest
frequency in the pre-sampled signal.
 The higher the value of L, the less distorted a
signal is recovered.

4.74
 Figure 4.27 Components of a PCM decoder

4.75
 Example 4.15

We have a low-pass analog signal of 4 kHz. If we send the

analog signal, we need a channel with a minimum
bandwidth of 4 kHz. If we digitize the signal and send 8
bits per sample, we need a channel with a minimum
bandwidth of 8 × 4 kHz = 32 kHz.

4.76
 Delta Modulation
 This scheme sends only the difference
between pulses, if the pulse at time tn+1 is
higher in amplitude value than the pulse at
time tn, then a single bit, say a “1”, is used to
indicate the positive value.
 If the pulse is lower in value, resulting in a
negative value, a “0” is used.
 This scheme works well for small changes in
signal values between samples.
 If changes in amplitude are large, this will
result in large errors.

4.77
 Figure 4.28 The process of delta modulation

4.78
 Figure 4.29 Delta modulation components

4.79
 Figure 4.30 Delta demodulation components

4.80
 Delta PCM (DPCM)

 Instead of using one bit to indicate positive

and negative differences, we can use more
bits -> quantization of the difference.
 Each bit code is used to represent the value
of the difference.
 The more bits the more levels -> the higher
the accuracy.

4.81
 2.2-3 TRANSMISSION MODES

The transmission of binary data across a link can be

accomplished in either parallel or serial mode. In
parallel mode, multiple bits are sent with each clock
tick. In serial mode, 1 bit is sent with each clock tick.
While there is only one way to send parallel data, there
are three subclasses of serial transmission:
asynchronous, synchronous, and isochronous.

Topics discussed in this section:

Parallel Transmission
Serial Transmission

4.82
 Figure 2.2.31 Data transmission and modes

4.83
 Figure 2.2.32 Parallel transmission

4.84
 Figure 2.2.33 Serial transmission

4.85
 Note

In asynchronous transmission, we send

1 start bit (0) at the beginning and 1 or
more stop bits (1s) at the end of each
byte. There may be a gap between
each byte.

4.86
 Note

Asynchronous here means

“asynchronous at the byte level,”
but the bits are still synchronized;
their durations are the same.

4.87
 Figure 2.2.34 Asynchronous transmission

4.88
 Note

In synchronous transmission, we send

bits one after another without start or
stop bits or gaps. It is the responsibility
of the receiver to group the bits.

4.89
 Figure 2.2.35 Synchronous transmission

4.90