2

DIGITAL TRANSMISSION
OVERVIEW

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TOPICS IN THIS SECTION

DIGITAL TRANSMISSION
PULSE CODE MODULATION
QUANTIZATION
TIME DIVISION MULTIPLEXING
EUROPEAN PRIMARY MULTIPLEXER
PLESIOCHRONOUS DIGITAL HIERARCHY
SYNCHRONOUS DIGITAL HIERARCHY
LINE CODING
ERROR DETECTION
CYCLIC REDUNDANCY CHECK
ERROR CORRECTION
FORWARD ERROR CORRECTION (REED SOLOMON/LDPC)

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DIGITAL TRANSMISSION
The medium used in digital transmission systems is usually designed so that it is only electrically stable in one of the two states,
equivalent to ‘on’ (binary value‘ l’) or ‘off’( binary value ‘0’). Thus a simple form of digital line system might use an electrical current
as the conveying medium, and control the current to fluctuate between two values, ‘current on’ and ‘current off’.

The principal advantage of digital over analogue transmission is the improved quality of connection. With only two ‘allowed’ states on
the line (‘off’ and ‘on’) it is not all that easy to confuse them even when the signal is distorted slightly along the line by electrical
noise or interference or some other cause. Digital line systems are thus relatively immune to interference. The receiving end only
needs to detect whether the received signal is above or below a given threshold value. If the pulse shape is not a ‘clean’ square shape,
it does not matter. Allow the same electrical disturbance to interfere with an analogue signal, and the result would be a low volume
crackling noise at the receiving end, which could well make the signal incomprehensible.

To make digital transmission still more immune to noise, it is normal practice to regenerate the signals at intervals along the line. A
regenerator reduces the risk of misinterpreting the received bit stream at the distant end of a long-haul line, by counteracting the
effects of attenuation and distortion, which show up in digital signals as pulse shape distortions. In this corrective function a digital
regenerator may be regarded as the equivalent of an analogue repeater.

A digital line system may be designed to run at almost any bit speed, but on a single digital circuit it is usually 64 kbit/s. This is
equivalent to a 4 kHz analogue telephone channel, as we shall see shortly. The bit speed of a digital line system is roughly equivalent
to the bandwidth of an analogue line system; the more information there is to be carried, the greater the required bit speed. Later in
the chapter we also discuss how individual 64 kbit/s digital channels can be multiplexed together on a single physical circuit via time
division multiplex (TDM).

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PCM
Pulse code modulation (PCM) functions by converting analogue 6
signals into a format compatible with digital transmission, and it
consists of four stages. 5
First, there is the translation of analogue electrical signals into
digital pulses. Second, these pulses are coded into a sequence 4

amplitude
suitable for transmission. Third, they are transmitted over the
digital medium. Fourth, they are translated back into the 3

analogue signal (or an approximation of it) at the receiving end.

2
PCM was invented as early as 1939, but it was only in the 1960s
that it began to be widely applied. This was mainly because 1
before the day of solid state electronics we did not have the
technology to apply the known principles of PCM effectively. 0
Speech or any other analogue signals are converted into a 0 1 2 3 4 5 6 7 8 9 10
time
sequence of binary digits by sampling the signal waveform at Figure-01
regular intervals. At each sampling instant the waveform Sampling a Waveform

amplitude is determined and, according to its magnitude, is

assigned a numerical value, which is then coded into its binary Decimal
Time Amplitude Numeric
form and transmitted over the transport medium. At there Value
receiving end, the original electrical signal is reconstructed by
0 0.00 0
translating it back from the incoming digitalized signal. The
technique is illustrated in Figure-01, which shows a typical 1 2.72 2

speech signal, with amplitude plotted against time. 2 1.25 1

Sampling is pre-determined to occur at intervals of time t
3 4.40 4
(usually measured in microseconds). The numerical values of the
sampled amplitudes, and their 8-bit binary translations, are 4 0.57 0
shown in Table-01. 5 2.20 2
Because the use of decimal points would make the business more
6 4.49 4
complex and increase the bandwidth required for transmission,
amplitude is represented by integer values only. When the 7 5.36 5
waveform amplitude does not correspond to an exact integer 8 4.25 4
value, as occurs at time 5 in Figure-01, an approximation is
9 2.20 2
made. Hence at 5, value 2 is used instead of the exact value of
2.20. This reduces the total number of digits that need to be 10 0.65 0
sent. Table-01
Waveform Samples from Figure-01

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PCM
The signal is reconstituted at the receiving end by generating a 6
stepped waveform, each step of duration t, with amplitude
according to the digit value received. The signal of Figure-01 is 5
therefore reconstituted as shown in Figure-02.
In the example, the reconstituted signal has a square waveform
rather than the smooth continuous form of the original signal. 4
This approximation affects the listener’s comprehension to an

amplitude
extent which depends on the amount of inaccuracy involved. The 3
similarity of the reconstituted signal to the original may be
improved by
-increasing the sampling rate (i.e. reducing the time 2
separation of samples) to increase the number of points on the
horizontal axis of Figure-01 at which samples are taken, and/or 1
-increasing the number of quantization levels (i.e.
wave amplitude levels). The quantization levels are the points on
the vertical scale of Figure-01. 0
0 1 2 3 4 5 6 7 8 9 10
time
Figure-02
Reconstruction of the waveform

However, without an infinite sampling rate and an infinite range of quantum values, it is impossible to match an original analogue
signal precisely. Consequently an irrecoverable element of quantization noise is introduced in the course of translating the original
analogue signal into its digital equivalent. The sampling rate and the number of quantization levels need to be carefully chosen to
keep this noise down to levels at which the received signal is comprehensible to the listener. The snag is that the greater the sampling
rate and the greater the number of quantization levels, the greater is the digital bit rate required to carry the signal. Here again a
parallel can be found with the bandwidth of an analogue transmission medium, where the greater is the required fidelity of an
analogue signal, the greater is the bandwidth required.

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PCM The minimum acceptable sampling rate for carrying a given analogue signal using digital
1

The Nyquist–Shannon Sampling Theorem

transmission is calculated according to a scientific principle known as the Nyquist Criterion(1).
The criterion states that the sample rate must be at least double the frequency of the analogue
signal being sampled. For a standard speech channel this equals 2x4kHz=8000 samples per
second, the normal bandwidth of a speech channel being 4 kHz.
The Nyquist–Shannon sampling theorem is a fundamental result in the field
of information theory, in particular telecommunications and signal
processing. Sampling is the process of converting a signal (for example, a
function of continuous time or space) into a numeric sequence (a function
of discrete time or space). The theorem states:

The number of quantization levels found (by subjective tests) to be appropriate for good speech
comprehension is 256. In binary digit (bit) terms this equates to an eight bit number, so that the
quantum value of each sample is represented by eight bits. The required transmission rate of a
digital speech channel is therefore 8000 samples per second, times 8 bits, or 64 kbit/s. In other
words a digital channel of 64 kbit/s capacity is equivalent to an analogue telephone channel
bandwidth of 4 kHz. This is the reason why the basic digital channel is designed to run at 64
If a function x(t) contains no frequencies higher than B cps, it is completely
determined by giving its ordinates at a series of points spaced 1/(2B)
seconds apart.
In essence the theorem shows that an analog signal that has been sampled
can be perfectly reconstructed from the samples if the sampling rate
exceeds 2B samples per second, where B is the highest frequency in the
original signal.

kbit/s. Channel capacity (Hartley – Shannon theorem)

Channel capacity as stated by Hartley’s law is, in the absence of noise:

C  2f. log 2 N
where
C = channel capacity, bits per second
δf = channel bandwidth, Hz
N = number of coding levels (2 in binary system)
When noise is present, the channel capacity calculated according to the
Hartley–Shannon theorem is:

C  f. log 2  1   S/N

S where= the ratio of total signal power
to total noise power at the receiver
 N  input within the bandwidth, δf .
QUANTIZATION
When the amplitude level at a sample point, does not exactly match one of the quantization levels, an approximation is made which
introduces what is called quantization noise. Now, if the 256 quantization levels were equally spaced over the amplitude range of the
analogue signal, then the low amplitude signals would incur far greater percentage quantization errors (and thus distortion) than
higher amplitude signals. For this reason, the quantization levels are not linearly spaced, but instead are more densely packed around
the zero amplitude level. This gives better signal quality in the low amplitude range and a more consistently clear signal across the
whole amplitude range. Two particular sets of quantization levels are in common use for speech signal quantization. They are called
the A-Law code and the μ-Law code. Both have a higher density of quantization levels around the zero amplitude level, and both use
an eight bit (256 level) coding technique. They only differ in the actual amplitude values chosen as their respective quantum levels.
The A-law code is the European standard for speech quantization and the μ-law code is used in North America. Unfortunately, because
the codes have non-corresponding quantization levels, conversion equipment is required for interworking and this adds to the
quantization noise of a connection comprising both A-law and μ-law digital transmission plant.

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TDM
As digital transmission is by discrete pulses and not continuous signals, it is possible for the information of more than one 64 kbit/s
channel to be transmitted on the same path, so long as the transmission rate (i.e. bit rate) is high enough to carry the bits from a
number of channels. In practice this is done by interleaving the pulses from the various channels in such a way that a sequence of
eight pulses from the first channel is followed by a sequence of eight from the second channel, and so on. The principle is illustrated
in Figure-3, in which the TDM equipment could be imagined to be a rotating switch, picking up in turn 8 bits (or 1 byte) from each of
the input channels A, B and C in turn. Thus the output bit stream of the TDM equipment is seen to comprise, in turn, byte A1 (from
channel A), byte B1 (from channel B), byte C1 (from channel C), then, cycling again, byte A2, byte B2, byte C2 ands o on. Note that a
higher bit rate is required on the output channel, to ensure that all the incoming data from all three channels can be transmitted
onward. As 3x2 = 6 bytes of data are received on the incoming side during a time period of 250 microseconds (1 byte on each channel
every 125 microseconds), all of them have to be transmitted on the outgoing circuit in an equal amount of time. As only a single
channel is used for output, this implies a rate of 6x8 = 48 bits in 250 microseconds, i.e. 192 kbit/s. (Unsurprisingly, the result is equal
to 3x64 kbit/s). Thus, in effect the various channels‘ time-share’ the outgoing transmission path.

A2 A1 1 Channel
Channel.A 3x64kbps=192kbps
00010110 01110110
(64kbps each)
3 Channels

B2 B1 C2 B2 A2 C1 B1 A1
Channel.B 01100101 01011111
01011111 11110001 00010110 10010000 11110001 01110110

C2 C1
Channel.C 48 bits transmitted in 250μs
01100101 10010000

16 bits @ 64kbps (250μs)

Figure-3
The Principle of Time Division Multiplexing

TDM can either be carried out by interleaving a byte (i.e. 8 bits) from each tributary channel in turn, or it can be done by single bit
interleaving. Figure-3 shows the more common method of byte interleaving.

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TDM
EUROPEAN PRIMARY MULTIPLEXER
The use of the TDM technique is so common on digital line
systems that physical circuits carrying only 64 kbit/s are
30 individual analogue circuits
extremely rare, so that digital line terminating equipment
(1-15 and 17-31)
usually includes a multiplexing function. Figure-4 shows a
64kbps digital data
typical digital line terminating equipment, used to
convert between a number of individual analogue 1 A/D
channels (carried on a number of individual physical 2048kbps digital data
2 A/D
circuits) and a single digital bit stream carried on a single
3 A/D
physical circuit. The equipment shown is called a primary
4 A/D TDM
multiplexer. A primary multiplexer (PMUX) contains an
Eqp.
analogue to digital conversion facility for individual
telephone channel conversion to 64 kbit/s, and 30 A/D
additionally a time division multiplex facility. In Figure-4
31 A/D
a PMUX of European origin is illustrated, converting 30
analogue channels into A-law encoded 64 kbit/s digital Analogue to digital signal conversion, using A-law
format, and then time division multiplexing all of these Pulse Code Modulation (PCM]
64 kbit/s channels into a single 2.048 Mbit/s (E1) digital
line system. (2.048 Mbit/s is actually equal to 32 X 64
Figure-4
kbit/s, but channels ‘0’ and ‘16’ of the European system European Primary Multiplexer
are generally used for purposes other than carriage of
information.)

The transmitting equipment of a digital line system has the job of multiplexing the bytes from all the constituent channels.
Conversely, the receiving equipment must disassemble these bytes in precisely the correct order. This requires synchronous operation
of transmitter and receiver, and to this end particular patterns of pulses are transmitted at set intervals, so that alignment and
synchronism can be maintained. These extra pulses are sent in channel 0 of the European 2 Mbit/s digital system, and in the extra
8kbit/s of the North American 1.5 Mbit/s system.

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TDM
PLESIOCHRONOUS DIGITAL HIERARCHY
The number of channels multiplexed on a carrier depends on the overall rate of bit transmission on the line. Given that each channel
must be transmitted at 64 kbits/s, the overall bit speed is usually related to an integer multiple of 64 kbit/s. There are three basic
hierarchies of transmission rates which have been standardized for international use, but these extend to higher bit rates than the
2.048 Mbit/s and 1.544 Mbit/s versions so far discussed.

The ITU-T( formerly CCITT, Consultative Committee for International Telephones and Telegraphs), CEPT (European Committee for
Posts and Telecommunications) and ETSI (European Telecommunications Standards Institute) have standardized 2.048 Mbit/s as the
primary digital bandwidth (E1 line system) and A-Law as the speech encoding algorithm.

This has 32 channels, 30 for speech and two for alignment synchronization and signalling, more of which we shall discuss later in the
chapter.

Higher transmission rates in the European digital hierarchy are attained by interleaving a number of 2 Mbit/s systems.
The standardized rates are:

2.048 Mbit/s referred to as E1 or 2 Mbit/s

8.448 Mbit/s referred to as E2 or 8 Mbit/s (4x2 Mbit/s)
34.368 Mbit/s referred to as E3 or 34 Mbit/s (4x8 Mbit/s)
139.264 Mbit/s referred to as E4 or 140 Mbit/s (4x34 Mbit/s)
564.992 Mbit/s referred to as E5 or 565 Mbit/s (4x140 Mbit/s)

PDH allows transmission of data streams that are nominally running at the same rate, but allowing some variation on the speed around
a nominal rate. By analogy, any two watches are nominally running at the same rate, clocking up 60 seconds every minute. However,
there is no link between watches to guarantee they run at exactly the same rate, and it is highly likely that one is running slightly
faster than the other.

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TDM
SYNCHRONOUS DIGITAL HIERARCHY
SDH (synchronous digital hierarchy) and SONET
(synchronous optical network) demand synchronous
operation of all the line systems within a network
(i.e. all must operate using the same clock). In
return, it offers a simpler and more regular frame
structure of 2Mbit/s and 1.5Mbit/s tributaries
within the higher bitrates (multiples of 155 Mbit/s).
This gives much greater flexibility in management
and administration of the system. A further
significant benefit is their support of both 2Mbit/s
and 1.5 Mbit/s based hierarchies, creating an easy
migration path for worldwide standardization.

The STM-1 (synchronous transport module level - 1)

frame is the basic transmission format for SDH or STM.1 Frame Structure
the fundamental frame or the first level of the
synchronous digital hierarchy. The STM-1 frame is
transmitted in exactly 125 microseconds, therefore
there are 8000 frames per second on a fiber-optic
circuit designated OC-3 (optical carrier three).
The STM-1 frame consists of overhead and pointers plus information payload. The first 9 columns of each frame make up the Section
Overhead and Administrative Unit Pointers, and the last 261 columns make up the Information Payload. The pointers (H1, H2, H3
bytes) identify administrative units (AU) within the information payload.

Carried within the information payload, which has its own frame structure of 9 rows and 261 columns, are administrative units
identified within the information payload by pointers. Within the administrative unit is one or more virtual containers (VC). VCs
contain path overhead and VC payload. The first column is for path overhead; it’s followed by the payload container, which can itself
carry other containers. Administrative units can have any phase alignment within the STM frame, and this alignment is indicated by
the pointer in row four,

The section overhead of an STM-1 signal (SOH) is divided into two parts: the regenerator section overhead (RSOH) and the multiplex
section overhead (MSOH). The overheads contain information from the system itself, which is used for a wide range of management
functions, such as monitoring transmission quality, detecting failures, managing alarms, data communication channels, service
channels, etc.

The STM frame is continuous and is transmitted in a serial fashion, byte-by-byte, row-by-row.

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Synchronous multiplex structure in compliance with ITU-T Recommendation G.707

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Overhead Bytes

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LINE CODING
The basic information to be transported over any digital line system, irrespective of its hierarchical level, is a sequence of ones and
zeros, also referred to as marks and spaces. The sequence is not usually sent directly to line, but is first arranged according to a line
code. This aids intermediate regenerator timing and distant end receiver timing, maximizing the possible regenerator separation and
generally optimizing the operation of the line system. The potential problem is that if either a long string of 0s or 1s were sent to line
consecutively then the line would appear to be either permanently ‘on’ or permanently ‘off’. Effectively a direct current condition is
transmitted to line. This is not advisable for two reasons. First the power requirement is increased and the attenuation is greater for
direct as opposed to alternating current. Second, any subsequent devices in the line cannot distinguish the beginning and end of each
individual bit. They cannot tell if the line is actually still ‘alive’. The problem gets worse as the number of consecutive 0s or 1s
increases. Line codes therefore seek to ensure that a minimum frequency of line state changes is maintained.

Figure-5 illustrates the most commonly used line codes. Generally they all seek to eliminate long sequences of 1s or 0s, and try to be
balanced codes, i.e. producing a net zero direct current voltage (thus the three state codes CM1 and HDB3 try to negate positive
pulses with negative ones). This reduces the problems of transmitting power across the line. The more sophisticated modern
techniques simultaneously seek to reduce the frequency of line state changes (the baud rate) so that higher user bitrates can be
carried.

The simplest line code illustrated in Figure-5 is a non-return to
zero (NRZ) code in which 1 means “on” and 0 means “off”. This
is perhaps the easiest to understand.
A return-to-zero (RZ) code is like NRZ except that marks return
to zero midway through the bit period, and not at the end of the
bit. Such coding has the advantage of lower required power and
constant mark pulse length in comparison with basic NRZ.
The length of the pulse relative to the total bit period is known
as the duty cycle. Synchronization and timing adjustment can
thus be achieved without affecting the mark pulse duration.
A variation of the NRZ and RZ codes is the CMI (coded mark
inversion) code recommended by ITU-T. In CMI, a 0 is
represented by the two signal amplitudes A1 and A2 which are
transmitted consecutively, each for half the bit duration. 1s are
sent as full bit duration pulses of one of the two line signal
amplitudes, the amplitude alternating between A1 and A2
between consecutive marks.
In the Manchester code, a higher pulse density helps to maintain
synchronization between the two communicating devices. Here
the transition from high-to-low represents a 1 and the reverse
transition (from low-to-high) a 0. The Manchester code is used in
ethernet LANs.
The AMI (alternate mark inversiona) nd HDB3 (high density
bipolar) codes defined by ITU-T (recommendation G.703) are
both three-state, rather than simple two-state (on/off) codes. In
these codes, as can be seen in Figure-5, the two extreme states
are used to represent marks, and the mid state is used to
represent spaces. The three states could be positive and negative
values, with a mid value of 0. In the case of optical fibres, where
light is used, the three states could be ‘off’, ‘low intensity’ and
‘high intensity’.
In both AMI and HDB3 line codes, alternative marks are sent as
positive and negative pulses. Alternating the polarity of the
pulses helps to prevent direct current being transmitted to line.
In a two-state code, a string of marks would have the effect of
sending a steady ‘on’ value to line.

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LINE CODING

1 0 1 1 0 0 0 0 1
The HDB3 code (used widely in Europe
and on international transmission
NRZ systems) is an extended form of AMI in
which the number of consecutive zeros
that may be sent to line is limited to
three. Limiting the number of consecutive
RZ zeros brings two benefits: first a null
signal is avoided, and second a minimum
mark density can be maintained (even
during idle conditions such as pauses in
CMI speech). A high mark density aids the
regenerator timing and synchronization.
In HDB3, the fourth zero in a string of
four is marked (i.e. forcibly set to 1) but
Manchester this is done in such a way that the ‘zero’
value of the original signal may be
recovered at the receiving end. The
recovery is achieved by marking fourth
AMI zeros in violation, that is to say, in the
same polarity as the previous ‘mark’,
rather than in opposite polarity mark
(opposite polarity of consecutive marks
V
HDB3 being the normal procedure).

Figure-5
Commonly used line codes for digital line systems

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ERROR DETECTION
CYCLIC REDUNDANCY CHECK
“A cyclic redundancy check (CRC) is a non-secure hash function designed to detect accidental changes to raw computer data, and is
commonly used in digital networks and storage devices such as hard disk drives. A CRC-enabled device calculates a short, fixed-length
binary sequence, known as the CRC code or just CRC, for each block of data and sends or stores them both together. When a block is
read or received the device repeats the calculation; if the new CRC does not match the one calculated earlier, then the block
contains a data error and the device may take corrective action such as rereading or requesting the block be sent again.”

TIMESLOT 0 This table, from G.704, shows the sequence of

FRAMES BIT.1 BIT.2 BIT.3 BIT.4 BIT.5 BIT.6 BIT.7 BIT.8 bits in the frame alignment (TS0) position of
0 C1 0 0 1 1 0 1 1 successive frames. In frames not containing
the frame alignment signal, the first bit is
SUBMULTIFRAME-1

1 0 1 A S S S S S
2 C2 0 0 1 1 0 1 1 used to transmit the CRC multi-frame signal
3 0 1 A S S S S S (001011) which defines the start of the sub-
4 C3 0 0 1 1 0 1 1 multi-frame. Alternate frames contain the
5 0 1 A S S S S S frame alignment word (0011011) preceded by
one of the Cyclic Redundancy Checksum-4
MULTIFRAME

6 C4 0 0 1 1 0 1 1
7 0 1 A S S S S S (CRC-4) bits. The CRC-4 remainder is
8 C1 0 0 1 1 0 1 1
calculated on all the 2048 bits of the previous
submulti-frame (SMF), and the 4-bit word sent
SUBMULTIFRAME-2

9 1 1 A S S S S S
as C1, C2, C3, C4 of the current SMF. (Note
10 C2 0 0 1 1 0 1 1
that the CRC-4 bits of the previous SMF are set
11 1 1 A S S S S S
to zero before the calculation is
12 C3 0 0 1 1 0 1 1
made.)
13 E 1 A S S S S S
At the receive end, the CRC remainder is
14 C4 0 0 1 1 0 1 1
recalculated for each SMF and the result
15 E 1 A S S S S S
compared with the CRC-4 bits received in the
E = CRC-4 error indication bits next SMF. If they differ, then it is assumed
C1 to C4 = Cyclic Redundancy Check-4 (CRC-4) bits that the checked SMF is in error. What this
S = Spare bits
A = Remote alarm indication
tells us is that a block of 2048 bits had one or
more errors.
Figure-6 (Refer to figure-7 for multiframe structure)

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MULTIFRAME.00 MULTIFRAME.01 MULTIFRAME.02 MULTIFRAME.03 MULTIFRAME.04

FRAME.00 FRAME.01 FRAME.02 FRAME.03 FRAME.04 FRAME.05 FRAME.06 FRAME.07 FRAME.08 FRAME.09 FRAME.10 FRAME.11 FRAME.12 FRAME.13 FRAME.14 FRAME.15

TS.00 TS.01 TS.02 TS.15 TS.16 TS.17 TS.30 TS.31

MFAS
TS.00 TS.01 TS.02 TS.15 TS.16 TS.17 TS.30 TS.31

FAS
NFAS
CAS

Figure-7
Multiframe structure of PCM.30

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ERROR CORRECTION
“Error detection is the ability to detect the presence of errors caused by noise or other impairments during transmission from the
transmitter to the receiver.
Error correction is the additional ability to reconstruct the original, error-free data.”

There are two basic ways to design the channel code and protocol for an error correcting system:

Automatic repeat-request (ARQ): The transmitter sends the data and also an error detection code, which the receiver uses to check
for errors, and requests retransmission of erroneous data. In many cases, the request is implicit; the receiver sends an
acknowledgement (ACK) of correctly received data, and the transmitter re-sends anything not acknowledged within a reasonable
period of time.
Forward error correction (FEC): The transmitter encodes the data with an error-correcting code (ECC) and sends the coded message.
The receiver never sends any messages back to the transmitter. The receiver decodes what it receives into the "most likely" data. The
codes are designed so that it would take an "unreasonable" amount of noise to trick the receiver into misinterpreting the data.
It is possible to combine the two, so that minor errors are corrected without retransmission, and major errors are detected and a
retransmission requested. The combination is called hybrid automatic repeat-request.

FORWARD ERROR CORRECTION

Forward error correction (FEC) code is redundant data that is
added to the message on the sender side. If the number of errors
is within the capability of the code being used, the receiver can
use the extra information to discover the locations of the errors
and correct them. Since the receiver does not have to ask the
sender for retransmission of the data, a back-channel is not
necessary in forward error correction, so it is suitable for simplex
communication such as broadcasting. Error-correcting codes are
used in computer data storage, for example CDs, DVDs and in
dynamic RAM. It is also used in digital transmission, especially
wireless communication, since wireless communication without
FEC often would suffer from packet-error rates close to 100%,
and conventional automatic repeat request error control would
yield a very low goodput.
Two main categories are convolutional codes and block codes.
Convolutional codes are processed on a bit-by-bit basis, and only
causes a processing delay corresponding to a few bit periods. In
convolutional coding, a Viterbi decoder is typically used on the
receiver side.
Block codes are processed on a block-by-block basis. Early
examples of block codes are repetition codes, Hamming codes
and multidimensional parity-check codes. More efficient codes,
often used in modern systems, are Reed-Solomon codes, turbo
codes, BCH codes, Reed-Muller codes, Binary Golay codes, and
low-density parity-check codes (LDPC). For example, the code
rate of the Reed Solomon block code denoted RS(204,188)
processes blocks of 188 bytes of useful information at a time,
and appends 204 - 188 = 16 redundant bytes to each block. It can
handle 8 incorrect bytes per data block.

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ERROR CORRECTION
Shannon's theorem is an important theorem in forward error correction which describes a theoretical upper bound for the attainable
spectral efficiency in bit/s/Hz of a channel coding scheme (an error-correcting scheme, typically combined width a digital modulation
scheme) for a specific signal-to-noise ratio (SNR). For example, if the SNR is 0 dB, the spectral efficiency can not be higher than 1
bit/s/Hz, resulting in that the information rate in bit/s (the useful bitrate, excluding redundant ECC) can not be higher than the
bandwidth in hertz.

The effectiveness of the coding scheme may also be measured in terms of code rate k/n, which is the ratio between k information bits
and n transmitted data bits. Finally the effectiveness may be expressed as coding gain in decibel, which is the allowable reduction in
signal-to-noise ratio whilst still attaining the same bit error rate and data rate as the uncoded system.

REED–SOLOMON ERROR CORRECTION

Reed–Solomon error correction is an error-correcting code that works by oversampling a polynomial constructed from the data. The
polynomial is evaluated at several points, and these values are sent or recorded. Sampling the polynomial more often than is necessary
makes the polynomial over-determined. As long as it receives "many" of the points correctly, the receiver can recover the original
polynomial even in the presence of a "few" bad points.

LOW-DENSITY PARITY-CHECK CODE

In information theory, a low-density parity-check code (LDPC code) is a linear error correcting code, a method of transmitting a
message over a noisy transmission channel, and is constructed using a sparse bipartite graph. LDPC codes are capacity-approaching
codes, which means that practical constructions exist that allow the noise threshold to be set very close (or even arbitrarily close on
the BEC) to the theoretical maximum (the Shannon limit) for a symmetric memoryless channel. The noise threshold defines an upper
bound for the channel noise up to which the probability of lost information can be made as small as desired. Using iterative belief
propagation techniques, LDPC codes can be decoded in time linear to their block length.

MW Transmission Network Planning & Design Training Digital Transmission

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