COMMUNICATIONS 2: MODULATION AND

CODING TECHNIQUES

Chapter 1
Introduction to Digital Communications Systems and
Digital Modulation

1.1. Elements of digital communication system

1.2. Introduction to digital modulation
1.3. Information Capacity, Bits, Bit Rate, Baud, and M-ary Encoding
1.4. Amplitude-Shift Keying
1.5. Frequency-Shift Keying
1.6. Phase-Shift Keying
1.7. Quadrature-Amplitude Modulation
1.8. Bandwidth Efficiency
1.9. Carrier Recovery
1.10. Clock Recovery
1.11. Differential Phase-Shift Keying
1.12. Trellis Code Modulation
1.13. Probability of Error and Bit Error Rate
1.14. Error Performance

After careful study of this chapter, the students will be able to:

1. Describe the elements of communication system and their functions

2. Define electronic communications, digital modulation and digital radio, digital communications
3. Define information capacity bit, bit rate, baud, and minimum bandwidth
4. Explain Shannon’s limit for information capacity and M-ary encoding
5. Define and describe digital amplitude modulation and frequency-shift keying
6. Describe continuous-phase frequency-shift keying
7. Define phase-shift keying
8. Explain binary phase-shift keying and quaternary phase-shift keying
9. Describe 8- and 16-PSK and quadrature-amplitude modulation
10. Explain 8-QAM and 16-QAM
11. Define bandwidth efficiency
12. Explain carrier recovery and clock recovery
13. Define and describe differential phase-shift keying
14. Define and explain trellis-code modulation
15. Define probability of error and bit error rate
16. Develop error performance equations for FSK, PSK, and QAM
1.1. Elements of digital communication system
 Figure 1.1 illustrates the functional diagram and the basic elements of a digital communication system.
 The source output may be either an analog signal, such as an audio or video signal, or a digital signal, such as the
output of a computer, that is discrete in time and has a finite number of output characters.
 In digital communication system, the messages produced by the source are converted into a sequence of binary
digits. Ideally, we should like to represent the source output (message) by as few binary digits as possible. In other
words, we seek an efficient representation of the source output that results in little or no redundancy.

Figure 1.1 Basic elements of a digital communication system

 The process of efficiently converting the output of either an analog or digital source into a sequence of binary digits
is called source encoding or data compression.
 The sequence of binary digits from the source encoder, which we call the information sequence, is passed to the
channel encoder. The purpose of the channel encoder is to introduce, in a controlled manner, some redundancy in
the binary information sequence that can be used at the receiver to overcome the effects of noise and interference
encountered in the transmission of the signal through the channel. Thus, the added redundancy serves to increase the
reliability of the received data and improves the fidelity of the received signal. In effect, redundancy in the
information sequence aids the receiver in decoding the desired information sequence.
 The binary sequence at the output of the channel encoder is passed to the digital modulator, which serves as the
interface to the communication channel. Since nearly all the communication channels encountered in practice are
capable of transmitting electrical signals (waveforms), the primary purpose of the digital modulator is to map the
binary information sequence into signal waveforms.
 The communication channel is the physical medium that is used to send the signal from the transmitter to the receiver.
In wireless transmission, the channel may be the atmosphere (free space). On the other hand, telephone channels
usually employ a variety of physical media, including wire lines, optical fiber cables, and wireless (microwave radio).
Whatever the physical medium used for transmission of the information, the essential feature is that the transmitted
signal is corrupted in a random manner by a variety of possible mechanisms, such as additive thermal noise generated
by electronic devices; man-made noise, e.g., automobile ignition noise; and atmospheric noise, e.g., electrical
lightning discharges during thunderstorms.
 At the receiving end of a digital communication system, the digital demodulator processes the channel-corrupted
transmitted waveform and reduces the waveforms to a sequence of numbers that represent estimates of the
transmitted data symbols (binary or M-ary). This sequence of numbers is passed to the channel decoder, which
attempts to reconstruct the original information sequence from knowledge of the code used by the channel encoder
and the redundancy contained in the received data.
 A measure of how well the demodulator and decoder perform is the frequency with which errors occur in the decoded
sequence. More precisely, the average probability of a bit-error at the output of the decoder is a measure of the
performance of the demodulator-decoder combination. In general, the probability of error is a function of the code
characteristics, the types of waveforms used to transmit the information over the channel, the transmitter power, the
characteristics of the channel (i.e., the amount of noise, the nature of the interference), and the method of
demodulation and decoding.
 As a final step, when an analog output is desired, the source decoder accepts the output sequence from the channel
decoder and, from knowledge of the source encoding method used, attempts to reconstruct the original signal from
the source.

1.2. Introduction to digital modulation

 Electronic communications is the transmission, reception, and processing of information with the use of electronic
circuits.
 Information is defined as knowledge or intelligence that is communicated (i.e., transmitted or received) between two
or more points.
  Digital modulation is the transmittal of digitally modulated analog signals (carriers) between two or more points in
a communications system. Digital modulation is sometimes called digital radio because digitally modulated signals
can be propagated through Earth’s atmosphere and used in wireless communications systems.
 Modern digital modulation systems offer several outstanding advantages over traditional analog systems, such as
ease of processing, ease of multiplexing, and noise immunity.
 Digital communications is a rather ambiguous term that could have entirely different meanings to different people.
In the context of this text, digital communications include systems where relatively high-frequency analog carriers
are modulated by relatively low frequency digital information signals (digital radio) and systems involving the
transmission of digital pulses (digital transmission). Digital transmission systems transport information in digital
form and, therefore, require a physical facility between the transmitter and receiver, such as a metallic wire pair, a
coaxial cable, or an optical fiber cable. In digital radio systems, the carrier facility could be a physical cable, or it
could be free space.
 The property that distinguishes digital radio systems from conventional analog modulation communications systems
is the nature of the modulating signal. Both analog and digital modulation systems use analog carriers to transport
the information through the system. However, with analog modulation systems, the information signal is also analog,
whereas with digital modulation, the information signal is digital, which could be computer generated data or digitally
encoded analog signals.
 Referring to Equation 1:

(1)

 If the information signal is digital and the amplitude (V) of the carrier is varied proportional to the information
signal, a digitally modulated signal called amplitude shift keying (ASK) is produced.
 If the frequency (f) is varied proportional to the information signal, frequency shift keying (FSK) is produced.
 If the phase of the carrier (θ) is varied proportional to the information signal, phase shift keying (PSK) is
produced.
 If both the amplitude and the phase are varied proportional to the information signal, quadrature amplitude
modulation (QAM) results. ASK, FSK, PSK, and QAM are all forms of digital modulation.
 Digital modulation is ideally suited to a multitude of communications applications, including both cable and wireless
systems. Applications include the following:
(1) relatively low-speed voice-band data communications modems, such as those found in most personal
computers; (2) high-speed data transmission systems, such as broadband digital subscriber lines (DSL);
(3) digital microwave and satellite communications systems; and
(4) cellular telephone Personal Communications Systems (PCS).

1.3. Information Capacity, Bits, Bit Rate, Baud, and M-ary Encoding
 Information theory is a highly theoretical study of the efficient use of bandwidth to propagate information through
electronic communications systems. Information theory can be used to determine the information capacity of a data
communications system.
 Information capacity is a measure of how much information can be propagated through a communications system
and is a function of bandwidth and transmission time. Information capacity represents the number of independent
symbols that can be carried through a system in a given unit of time.
 The most basic digital symbol used to represent information is the binary digit, or bit. Therefore, it is often convenient
to express the information capacity of a system as a bit rate.
 Bit rate is simply the number of bits transmitted during one second and is expressed in bits per second (bps).
 Hartley of Bell Telephone Laboratories developed that information capacity is a linear function of bandwidth and
transmission time and is directly proportional to both. If either the bandwidth or the transmission time changes, a
directly proportional change occurs in the information capacity. Simply stated, Hartley’s law is
𝐼 ∝𝐵×𝑡 (2)

where I = information capacity (bits per second)

B = bandwidth (hertz)
t = transmission time (seconds)

 Claude E. Shannon (also of Bell Telephone Laboratories) published a paper in the Bell System Technical Journal
relating the information capacity of a communications channel to bandwidth and signal-to-noise ratio. The higher
the signal-to-noise ratio, the better the performance and the higher the information capacity. Mathematically stated,
the Shannon limit for information capacity is
𝑆
𝐼 = 𝐵 log 2 (1 + ) (3)
𝑁

𝑆
or 𝐼 = 3.32𝐵 log10 (1 + ) (4)
𝑁
where I = information capacity (bps)
B = bandwidth (hertz)
𝑆
= signal-to-noise power ratio (unitless)
𝑁
Examples:
1. Calculate the capacity of a standard 4 kHz telephone channel with a 32-dB signal-to-noise ratio. (323,953 bps)
2. A system has a bandwidth of 4 kHz and a signal-to-noise ratio of 28 dB at the input to the receiver. Calculate
(a) its information-carrying capacity (37,216 bps)
(b) the capacity of the channel if its bandwidth is doubled, while the transmitted signal power remains constant.
(66,448 bps)
3. Calculate the maximum channel capacity of a voice-grade telephone line with a bandwidth of 3100 Hz and an S/N of
30-dB. (31,000 bps)
4. The bandwidth of a communication channel is 12.5 kHz. The S/N ratio is 25-dB. Calculate the maximum theoretical
channel capacity. (103,805.3 bps)

 M-ary encoding: M-ary is a term derived from the word binary. M simply represents a digit that corresponds to the
number of conditions, levels, or combinations possible for a given number of binary variables. It is often
advantageous to encode at a level higher than binary (sometimes referred to as beyond binary or higher-than-binary
encoding) where there are more than two conditions possible. For example, a digital signal with four possible
conditions (voltage levels, frequencies, phases, and so on) is an M-ary system where M = 4. If there are eight possible
conditions, M = 8 and so forth.
 The number of bits necessary to produce a given number of conditions is expressed mathematically as
𝑁 = log 2 𝑀 (5)
where N = number of bits necessary
M = number of conditions, levels, or combinations possible with N bits
 The number of conditions possible with N bits is expressed mathematically as
2𝑁 = 𝑀 (6)
 Baud is a term that is often misunderstood and commonly confused with bit rate (bps). Bit rate refers to the rate of
change of a digital information signal, which is usually binary. Baud, like bit rate, is also a rate of change; however,
baud refers to the rate of change of a signal on the transmission medium after encoding and modulation have occurred.
Hence, baud is a unit of transmission rate, modulation rate, or symbol rate and, therefore, the terms symbols per
second and baud are often used interchangeably. Mathematically, baud is the reciprocal of the time of one output
signaling element, and a signaling element may represent several information bits. Baud is expressed as
1
𝑏𝑎𝑢𝑑 = (7)
𝑡𝑠
where baud = symbol rate (baud per second)
ts = time of one signaling element (seconds)
 A signaling element is sometimes called a symbol and could be encoded as a change in the amplitude, frequency, or
phase.
 A baud is transmitted one at a time; however, a baud may represent more than one information bit. Thus, the baud of
a data communications system may be considerably less than the bit rate.
 According to H. Nyquist, binary digital signals can be propagated through an ideal noiseless transmission medium at
a rate equal to two times the bandwidth of the medium. The minimum theoretical bandwidth necessary to propagate
a signal is called the minimum Nyquist bandwidth or sometimes the minimum Nyquist frequency. Thus,
𝑓𝑏 = 2𝐵 (8)
where fb = bit rate in bps and
B = ideal Nyquist bandwidth.
 The actual bandwidth necessary to propagate a given bit rate depends on several factors, including the type of
encoding and modulation used, the types of filters used, system noise, and desired error performance. The ideal
bandwidth is generally used for comparison purposes only.
 Using multilevel signaling, the Nyquist formulation for channel capacity is
𝑓𝑏 = 2𝐵 log 2 𝑀 (9)
where fb = channel capacity (bps)
B = minimum Nyquist bandwidth (hertz)
M = number of discrete signal or voltage levels
Rearranging equation (9), the minimum bandwidth necessary to pass M-ary digitally modulated carriers is
𝑓 𝑓
𝐵=( 𝑏 )= 𝑏 (10)
log2 𝑀 𝑁
where N = number of bits encoded into each signaling element.
If information bits are encoded (grouped) and then converted to signals with more than two levels, transmission rates
in excess of 2B are possible. In addition, since baud is the encoded rate of change, it also equals the bit rate divided
by the number of bits encoded into one signaling element. Thus,
𝑓
𝑏𝑎𝑢𝑑 = 𝑏 (11)
𝑁
By comparing Equation 10 with Equation 11, it can be seen that with digital modulation, the baud and the ideal
minimum Nyquist bandwidth have the same value and are equal to the bit rate divided by the number of bits encoded.
This statement holds true for all forms of digital modulation except frequency-shift keying.
1.4. Amplitude-Shift Keying
 Amplitude-Shift Keying (ASK) is the simplest digital modulation technique, where a binary information signal directly
modulates the amplitude of an analog carrier.
ASK is similar to standard amplitude modulation except there are only two output amplitudes possible. Amplitude-
shift keying is sometimes called digital amplitude modulation (DAM). Mathematically, amplitude-shift keying is

𝐴
𝑣𝑎𝑠𝑘 (𝑡) = [1 + 𝑣𝑚 (𝑡)] [ cos(𝜔𝑐 𝑡)] (12)
2

where vask(t) = amplitude-shift keying wave

vm(t) = digital information (modulating) signal (volts)
A/2 = unmodulated carrier amplitude (volts)
ωc = analog carrier radian frequency (radians per second, 2π fct)

In Equation 12, the modulating signal [vm(t)] is a normalized binary waveform, where +1 V = logic 1 and -1 V =
logic 0. Therefore, for a logic 1 input, vm(t) = +1 V, Equation 12 reduces to

𝐴
𝑣𝑎𝑠𝑘 (𝑡) = [1 + 1] [ cos(𝜔𝑐 𝑡)]
2

𝑣𝑎𝑠𝑘 (𝑡) = 𝐴 cos(𝜔𝑐 𝑡)

and for a logic 0 input, vm(t) = -1 V, Equation 12 reduces to

𝐴
𝑣𝑎𝑠𝑘 (𝑡) = [1 − 1] [ cos(𝜔𝑐 𝑡)]
2

𝑣𝑎𝑠𝑘 (𝑡) = 0
Thus, the modulated wave vask(t), is either A=cos(ωct) or 0. Hence, the carrier is either “on” or “off,” which is why
amplitude-shift keying is sometimes referred to as on-off keying (OOK).

ASK is used in some types of data communication when binary information is to be transmitted.

 Figure 1.4.1 shows the block diagram for the generation of ASK signal. ASK modulator is essentially an analog
multiplier that takes baseband message vm and passband carrier vc and multiplies the two resulting in the product
signal termed as ASK.

Figure 1.4.1 Block diagram of generation of ASK signal

 Figure 1.4.2 shows the time domain representation of the generation of ASK signal. The digital message i.e., binary
sequence can be represented as a message signal as shown in Fig. 1.4.2a. The carrier signal of frequency fc is generated
continuously from an oscillator circuit as shown in Fig. 1.4.2b. When the oscillator output is multiplied by the
message signal, it results in a signal as shown in Fig. 1.4.2c termed as ASK signal. When the binary symbol is one,
the ASK signal will have information equal to the carrier multiplied by message amplitude and when the binary
symbol is 0, it will be zero. Thus the output shifts between two amplitude levels, namely, Vm, Vc and 0. Hence the
name amplitude-shift keying.
From Figure 1.4.2, it can be seen that the time of one bit (tb) equals the time of one analog signaling element (ts). The
bit time is the reciprocal of the bit rate and the time of one signaling element is the reciprocal of the baud. Therefore,
the rate of change of the ASK waveform (baud) is the same as the rate of change of the binary input (bps); thus, the
bit rate equals the baud. With ASK, the bit rate is also equal to the minimum Nyquist bandwidth. This can be verified
by substituting into Equations 10 and 11 and setting N to 1:
𝑓𝑏 𝑓𝑏
𝐵= = 𝑓𝑏 𝑏𝑎𝑢𝑑 = = 𝑓𝑏
1 1
 Figure 1.4.2 Time domain representation of generation of ASK signal: (a) message, (b) carrier, and (c) ASK signal

1.5. Frequency-Shift Keying

 Frequency-shift keying (FSK) is another relatively simple, low-performance type of digital modulation. FSK is a
form of constant-amplitude angle modulation similar to standard frequency modulation (FM) except the modulating
signal is a binary signal that varies between two discrete voltage levels rather than a continuously changing analog
waveform. Consequently, FSK is sometimes called binary FSK (BFSK). The general expression for FSK is

𝑣𝑓𝑠𝑘 (𝑡) = 𝑉𝑐 cos{2𝜋[𝑓𝑐 + 𝑣𝑚 (𝑡)∆𝑓]𝑡} (13)

where vfsk(t) = binary FSK waveform

Vc = peak analog carrier amplitude (volts)
fc = analog carrier center frequency (hertz)
Δf = peak change (shift) in the analog carrier frequency (hertz)
vm(t) = binary input (modulating) signal (volts)

From Equation 13, it can be seen that the peak shift in the carrier frequency (Δf) is proportional to the amplitude of
the binary input signal (vm[t]), and the direction of the shift is determined by the polarity. The modulating signal is a
normalized binary waveform where a logic 1 = +1 V and a logic 0 = -1 V. Thus, for a logic 1 input, vm(t) = +1,
Equation 13 can be rewritten as

𝑣𝑓𝑠𝑘 (𝑡) = 𝑉𝑐 cos{2𝜋[𝑓𝑐 + ∆𝑓]𝑡}

For a logic 0 input, vm(t) = -1, Equation 13 becomes

𝑣𝑓𝑠𝑘 (𝑡) = 𝑉𝑐 cos{2𝜋[𝑓𝑐 − ∆𝑓]𝑡}

 FSK is widely used in the transmission of binary data in Bluetooth headsets, wireless speakers, and many forms of
industrial wireless.

 With binary FSK, the carrier center frequency (fc) is shifted (deviated) up and down in the frequency domain by the
binary input signal as shown in Figure 3. As the binary input signal changes from a logic 0 to a logic 1 and vice versa,
the output frequency shifts between two frequencies: a mark, or logic 1 frequency (fm), and a space, or logic 0
frequency (fs). The mark and space frequencies are separated from the carrier frequency by the peak frequency
deviation (Δf) and from each other by 2 Δf. Frequency deviation is illustrated in Figure 3 and expressed
mathematically as
|𝑓 −𝑓 |
∆𝑓 = 𝑚 𝑠 (14)
2
where Δf = frequency deviation (hertz)
|𝑓𝑚 − 𝑓𝑠 | = absolute difference between the mark and space frequencies (hertz)

 Referring to Figure 1.5.1a, the mark frequency is the higher frequency (fc + Δf), and the space frequency is the lower
frequency (fc −Δf), although this relationship could be just the opposite. In Figure 1.5.1b, the truth table shows the
input and output possibilities for a given digital modulation scheme.
 Figure 1.5.1 FSK in the time domain: (a)waveform; (b) truth table

 The baud for binary FSK can also be determined by substituting N = 1 in Equation 11:
𝑓𝑏
𝑏𝑎𝑢𝑑 = = 𝑓𝑏
1
 FSK is the exception to the rule for digital modulation, as the minimum bandwidth is not determined from Equation
10. The minimum bandwidth for FSK is given as
𝐵 = |(𝑓𝑠 + 𝑓𝑏 ) − (𝑓𝑚 − 𝑓𝑏 )|
= |𝑓𝑠 − 𝑓𝑚 | + 2𝑓𝑏
= 2𝛥𝑓 + 2𝑓𝑏
𝐵 = 2(𝛥𝑓 + 𝑓𝑏 ) (15)
where B = minimum Nyquist bandwidth (hertz)
Δf = frequency deviation (|fm - fs|) (hertz)
fb = input bit rate (bps)
 The formula used for modulation index in FM is also valid for FSK; thus,
∆𝑓
ℎ= (16)
𝑓𝑎
or
|𝑓𝑠 −𝑓𝑚|
ℎ= (17)
𝑓𝑏

where h = FM modulation index called the h-factor in FSK

𝑓
fa = fundamental frequency of the binary modulating signal (hertz) ,( 𝑏 )
2
fb = input bit rate (bps)
Δf = peak frequency deviation (hertz)
fm = mark frequency (hertz)
fs = space frequency (hertz)

 Figure 1.5.2 shows a simplified binary FSK modulator, which is very similar to a conventional FM modulator and is
very often a voltage-controlled oscillator (VCO). The center frequency (fc) is chosen such that it falls halfway between
the mark and space frequencies. A logic 1 input shifts the VCO output to the mark frequency, and a logic 0 input
shifts the VCO output to the space frequency. Consequently, as the binary input signal changes back and forth
between logic 1 and logic 0 conditions, the VCO output shifts or deviates back and forth between the mark and space
frequencies.
 A VCO-FSK modulator can be operated in the sweep mode where the peak frequency deviation is simply the product
of the binary input voltage and the deviation sensitivity of the VCO. With the sweep mode of modulation, the
frequency deviation is expressed mathematically as
∆𝑓 = 𝑣𝑚 (𝑡)𝑘𝑙

where Δf = peak frequency deviation (hertz)

vm(t) = peak binary modulating-signal voltage (volts)
kl = deviation sensitivity (hertz per volt).

 With binary FSK, the amplitude of the input signal can only be one of two values, one for a logic 1 condition and
one for a logic 0 condition. Therefore, the peak frequency deviation is constant and always at its maximum value.
Frequency deviation is simply plus or minus the peak voltage of the binary signal times the deviation sensitivity of
the VCO. Since the peak voltage is the same for a logic 1 as it is for a logic 0, the magnitude of the frequency
deviation is also the same for a logic 1 as it is for a logic 0.
 Figure 1.5.2 FSK modulator

 FSK demodulation is quite simple with a circuit such as the one shown in Figure 1.5.3. This type of FSK detection
is referred to as noncoherent detection; there is no frequency involved in the demodulation process that is
synchronized either in phase, frequency, or both with the incoming FSK signal. The FSK input signal is
simultaneously applied to the inputs of both bandpass filters (BPFs) through a power splitter. The respective filter
passes only the mark or only the space frequency on to its respective envelope detector. The envelope detectors, in
turn, indicate the total power in each passband, and the comparator responds to the largest of the two powers.

Figure 1.5.3 Noncoherent FSK demodulator

 Figure 1.5.4 shows the block diagram for a coherent FSK receiver. The incoming FSK signal is multiplied by a
recovered carrier signal that has the exact same frequency and phase as the transmitter reference. However, the two
transmitted frequencies (the mark and space frequencies) are not generally continuous; it is not practical to reproduce
a local reference that is coherent with both of them. Consequently, coherent FSK detection is seldom used.

Figure 1.5.4 Coherent FSK demodulator

 The most common circuit used for demodulating binary FSK signals is the phase locked loop (PLL), which is shown
in block diagram form in Figure 1.5.5. A PLL-FSK demodulator works similarly to a PLL-FM demodulator. As the
input to the PLL shifts between the mark and space frequencies, the dc error voltage at the output of the phase
comparator follows the frequency shift. Because there are only two input frequencies (mark and space), there are also
only two output error voltages. One represents a logic 1 and the other a logic 0.

Figure 1.5.5 PLL FSK demodulator

 Binary FSK has a poorer error performance than PSK or QAM and, consequently, is seldom used for high-
performance digital radio systems. Its use is restricted to low-performance, low-cost, asynchronous data modems that
are used for data communications over analog, voice-band telephone lines.
Examples:
1. Determine the bandwidth and baud for an FSK signal with a mark frequency of 32 kHz, a space frequency
of 24 kHz, and a bit rate of 4 kbps. (16 kHz, 4000 baud)
2. Determine the maximum bit rate for an FSK signal with a mark frequency of 48 kHz, a space frequency of
52 kHz, and an available bandwidth of 10 kHz. (3 kbps)
3. Determine the bandwidth and baud for an FSK signal with a mark frequency of 99 kHz, a space frequency
of 101 kHz, and a bit rate of 10 kbps. (22 kHz, 10 kbaud)
4. Determine the maximum bit rate for an FSK signal with a mark frequency of 102 kHz, a space frequency of
104 kHz, and an available bandwidth of 8 kHz. (3 kbps)