Digital Communication 3

Objectives
After studying this chapter, you should be able to:
( Compare analog and digital communication techniques and discuss the
advantages of each.
( Calculate the minimum sampling rate for a signal and explain the necessity
for sampling at that rate or above.
( Find the spurious frequencies produced by aliasing when the sample rate is
too low.
( Describe the common types of analog pulse modulation.
( Describe pulse-code modulation and calculate the number of quantizing
levels, the bit rate, and the dynamic range for PCM systems.
( Explain companding, show how it is accomplished, and explain its effects.
( Describe the coding and decoding of a PCM signal.
( Describe differential PCM and explain its operation and advantages.
( Describe delta modulation and explain the advantages of adaptive delta
modulation.
( Distinguish between lossless and lossy compression and provide examples
of each.
( Describe the operation of common types of vocoders.
84 ! CHAPTER 3

' 3.1 Introduction

In the previous chapter we looked at analog modulation techniques. These
were historically the earliest ways to transmit information by radio, and
they are still very popular. However, recent developments have made digital
methods more important, to the point where it is expected that most new
wireless communication systems will be digital.
In this chapter we will look at the digital encoding of analog signals.
Voice is a good example of such a signal; it begins and ends as analog infor-
mation. Once coded digitally, the analog signal is indistinguishable from
any other data stream, such as a word-processing document. In Chapter 4 we
will look at the transmission of this data by radio.
While the advantage of digital transmission for data is self-evident, it is
not immediately obvious why it is desirable to go to the trouble and expense
of converting analog signals to digital form and back again. There are in fact
several very good reasons for doing so. Here are some of the most important
for wireless communication systems.
( Digital signals can be manipulated more easily than analog signals.
They are easier to multiplex, for instance.
( Digital signals can easily be encrypted to ensure privacy.
( When an analog signal goes through a chain of signal processors,
such as transmitters, receivers, and amplifiers, noise and distortion
accumulate. This process can be made much less severe in digital sys-
tems by regenerating the signal and by using various types of error
control. These concepts will be explained in more detail later in this
chapter.
( Data compression can be used with a digital signal to reduce its band-
width to less than that required to transmit the original analog signal.
Until recently the use of digital techniques was restricted because dig-
ital communication required more bandwidth than analog. That situ-
ation has changed quite radically in recent years.
You might have noticed that “higher fidelity” and “improved frequency
response” do not appear in the above list. It is certainly true that some digi-
tal schemes, such as compact disc audio, have better fidelity than some ana-
log schemes, like FM broadcasts; but this is not always the case. The
distortion and frequency response of a digital audio system are fixed by the
sampling process, which will be described in the next section. The quality of
the reproduction can be predicted quite accurately, and the use of regenera-
tion and error correction can prevent it from deteriorating very much, but
not all digital transmission is of high quality. For instance, whether you
know it or not you have been communicating digitally for years: practically
 DIGITAL COMMUNICATION ! 85

all modern telephone switches are digital. Digital telephony sounds pretty
good, for a telephone call, but does not compare with compact disc audio.
We will find out why in the next sections.

' 3.2 Sampling

An analog signal varies continuously with time. If we want to transmit such
a signal digitally, that is, as a series of numbers, we must first sample the sig-
nal. This involves finding its amplitude at discrete time intervals. Only in
this way can we arrive at a finite series of numbers to transmit.

Sampling Rate In 1928, Harry Nyquist showed mathematically that it is possible to recon-
struct a band-limited analog signal from periodic samples, as long as the
sampling rate is at least twice the frequency of the highest-frequency com-
ponent of the signal. This assumes that an ideal low-pass filter prevents
higher frequencies from entering the sampler. Since real filters are not ideal,
in practice the sampling rate must be considerably more than twice the max-
imum frequency to be transmitted.
If the sampling rate is too low, a form of distortion called aliasing
or foldover distortion is produced. In this form of distortion, frequencies
in the sampled signal are translated downward. Figure 3.1 shows what
happens.
In Figure 3.1(a) the sampling rate is adequate and the signal can be re-
constructed. In Figure 3.1(b), however, the rate is too low and the attempt to
reconstruct the original signal results in a lower-frequency output signal.
Once aliasing is present, it cannot be removed.
The frequency of the interference generated by aliasing is easier to see by
looking at the frequency domain. For simplicity, assume that the signal to be
sampled, which we will call the baseband signal, is a sine wave:

e b = E b sin ω b t (3.1)
where
e b = instantaneous baseband signal voltage
E b = peak baseband signal voltage
ω b = the radian frequency of the baseband signal

The sampling process is equivalent to multiplying the baseband signal

by a pulse train at the sampling frequency. See Figure 3.2. For simplicity, as-
sume that the pulses have an amplitude of one volt. Then, when a pulse is
86 ! CHAPTER 3

FIGURE 3.1 Aliasing

present, the output of the sampler will be the same as the baseband signal
amplitude, and when there is no pulse, the sampler output will be zero.
The spectrum for the pulse train in Figure 3.2 is given in Chapter 1 as:
1 1  sin πτ/ T sin 2πτ/ T sin 3πτ/ T 
es = +2  cos ω st + cos 2ω st + cos 3ω st + ⋅ ⋅ ⋅ (3.2)
T T  πτ/ T 2πτ/ T 3πτ/ T 
1 2 sin πτ/ T sin 2πτ/ T 2 sin 3πτ/ T
= + cos ω st + cos 2ω st + cos 3ω st + ⋅ ⋅ ⋅
T πτ πτ 3πτ

where
es = instantaneous voltage of the sampling pulse
τ = pulse duration
T = pulse period
ωs = radian frequency of the pulse train
 DIGITAL COMMUNICATION ! 87

FIGURE 3.2
Sampling pulses

Multiplying the two signals given in Equations (3.1) and (3.2) together
gives the following output:

Eb 2 E b sin πτ / T
v (t ) = sin ω bt + sin ω bt cos ω s t (3.3)
T πτ

E b sin 2 πτ / T
+ sin ω bt cos 2ω s t
πτ

2 E b sin 3πτ / T
+ sin ω bt cos 3ω s t
3πτ
As is often the case with equations of this type, we do not have to “solve”
anything to understand what is happening. The first term is simply the origi-
nal baseband signal multiplied by a constant. If only this term were present,
the original signal could be recovered from the sampled signal. However, we
need to look at the other terms to see whether they will interfere with the
signal recovery.
The second term contains the product of sin ωbt and cos ωst. Recall the
trigonometric identities:

1
sin A cos B =
2
[sin( A − B) + sin( A + B)] (3.4)

and

sin( − A) = − sin A (3.5)

Equation (3.5) can be used to express Equation (3.4) in a more convenient

form for the problem at hand. Substituting Equation (3.5) into Equation
(3.4) gives:

1
sin A cos B =
2
[sin( B + A) − sin( B − A)] (3.6)
88 ! CHAPTER 3

This new identity can be used to expand the second term of Equation
(3.3) as follows:

2 E b sin πτ / T
sin ω bt cos ω s t (3.7)
πτ
E b sin πτ / T
=
πτ
[sin(ω s + ω b )t − sin(ω s − ω b )t ]
E b sin πτ / T E sin πτ / T
= sin(ω s + ω b )t − b sin(ω s − ω b )t
πτ πτ
Now we can see that this term consists of components at the sum and
difference of the baseband and sampling frequencies. The sum term can
easily be eliminated by a low-pass filter, since its frequency is obviously
much higher than the baseband. The difference term is more interesting. If
ws > 2ωb, then ωs − ωb > ωb and the difference part of this term can also be re-
moved by a low-pass filter, at least in theory. However, if ωs < 2ωb, the differ-
ence will be less than ωb. An aliased component will appear as

ƒa = ƒs − ƒb (3.8)
and low-pass filtering will not be effective in removing it.
The other terms in Equation (3.3) are not interesting here because they
all represent frequencies greater than ωs − ωb. Therefore, if we make sure that
(ωs − ωb) > ωb, these other terms will not be a problem. Let us, then, rewrite
Equation (3.3), including only the first term and the expanded second term:

Eb E sin πτ / T
v (t ) = sin ω bt + b sin(ω s + ω b )t (3.9)
T πτ
E b sin πτ / T
− sin(ω s − ω b )t
πτ
An example will further clarify the problem.

EXAMPLE 3.1 Y
A digital communication system uses sampling at 10 kilosamples per second
(kSa/s). The receiver filters out all frequencies above 5 kHz. What frequen-
cies appear at the receiver for each of the following signal frequencies at the
input to the transmitter?
(a) 1 kHz
(b) 5 kHz
(c) 6 kHz
 DIGITAL COMMUNICATION ! 89

SOLUTION
(a) The first term in Equation (3.9) is simply the input frequency, which is,
of course, the only one we want to see in the output. The second term
is the sum of the input and the sampling frequencies, and the third is the
difference. In this case, the frequencies generated are:

ƒb = 1 kHz ƒs + ƒb = 11 kHz ƒs − ƒb = 9 kHz

However, only the 1-kHz component passes through the filter and the sys-
tem operates correctly.
(b) ƒb = 5 kHz ƒs + ƒb = 15 kHz ƒs − ƒb = 5 kHz

Again, the system works properly and the 15-kHz component is removed
by the filter, and only the input frequency of 5 kHz appears at the output.
(c) ƒb = 6 kHz ƒs + ƒb = 16 kHz ƒs − ƒb = 4 kHz

Here we have a serious problem. The 16-kHz component is removed by

the filter but the 4-kHz component is not. With 6 kHz at the input we pro-
duce 6 kHz and 4 kHz at the output.
X

Natural and The equations in the previous section assumed that a sample consisted of
Flat-Topped the baseband signal multiplied by a rectangular pulse. To simplify the math-
Sampling ematics, we assumed that the pulse had an amplitude of 1 V. These assump-
tions yield a sample pulse whose shape follows that of the original signal, as
shown in Figure 3.3(a). This technique is called natural sampling.

FIGURE 3.3 Natural and flat-topped sampling

90 ! CHAPTER 3

Practical systems generally sample by using a sample-and-hold circuit,

which maintains the signal level at the start of the sample pulse. The results
of such a method are shown in Figure 3.3(b). This technique is known as
flat-topped sampling. If the samples were to be transmitted directly as ana-
log pulses of different amplitudes, there would be some small difference in
frequency response between the two techniques. However, for digital trans-
mission there is no practical difference between the two sampling methods.

Analog Pulse As previously mentioned, it would be possible to transmit the samples di-
Modulation rectly as analog pulses. This technique, called pulse-amplitude modulation
(PAM), does not offer any great advantage over conventional analog trans-
mission. In current systems, PAM is used as an intermediate step; before be-
ing transmitted, the PAM signal is digitized. Similarly, at the receiver, the
digital signal is converted back to PAM as part of the demodulation process.
The original signal can then be recovered using a low-pass filter.
Some improvement in noise performance can be made by transmitting
pulses of equal amplitude but variable length (with the duration of the
pulses corresponding to the amplitude of the samples). This technique is
called pulse-duration modulation (PDM) or pulse-width modulation
(PWM). PDM has uses in communication, in some telemetry systems for in-
stance; but it is not likely to be seen in modern wireless systems. Similarly, it
is possible to transmit the information signal by using pulses of equal ampli-
tude and duration but changing their timing in accordance with the sample
amplitude. This system, called pulse-position modulation (PPM), is men-
tioned only for completeness, as it is rarely seen. Figure 3.4 shows the basic
nature of all these systems.

' 3.3 Pulse-Code Modulation

Pulse-code modulation (PCM) is the most commonly used digital modula-
tion scheme. In PCM the available range of signal voltages is divided into
levels, and each is assigned a binary number. Each sample is then repre-
sented by the binary number representing the level closest to its amplitude,
and this number is transmitted in serial form. In linear PCM, levels are sepa-
rated by equal voltage gradations.

Quantization and The number of levels available depends on the number of bits used to ex-
Quantizing Noise press the sample value. The number of levels is given by

N = 2m (3.10)
 DIGITAL COMMUNICATION ! 91

FIGURE 3.4 Analog pulse modulation

where
N = number of levels
m = number of bits per sample

EXAMPLE 3.2 Y
Calculate the number of levels if the number of bits per sample is:
(a) 8 (as used in telephony)
(b) 16 (as used in the compact disc audio system)

SOLUTION
(a) The number of levels with 8 bits per sample is, from Equation (3.10),
N = 2m
= 28
= 256
92 ! CHAPTER 3

(b) The number of levels with 16 bits per sample is, from the same equation,

N = 2m
= 216
= 65536
X

This process is called quantizing. Since the original analog signal can
have an infinite number of signal levels, the quantizing process will produce
errors called quantizing errors or often quantizing noise.
Figure 3.5 shows how quantizing errors arise. The largest possible error is
one-half the difference between levels. Thus the error is proportionately
greater for small signals. This means that the signal-to-noise ratio varies
with the signal level and is greatest for large signals. The level of quantizing
noise can be decreased by increasing the number of levels, which also in-
creases the number of bits that must be used per sample.
The dynamic range of a system is the ratio between the strongest possi-
ble signal that can be transmitted and the weakest discernible signal. For
a linear PCM system, the maximum dynamic range in decibels is given
approximately by

DR = 1.76 + 6.02m dB (3.11)

where
DR = dynamic range in decibels
m = number of bits per sample
This equation ignores any noise contributed by the analog portion of
the system.

FIGURE 3.5 Quantizing error

 DIGITAL COMMUNICATION ! 93

EXAMPLE 3.3 Y
Find the maximum dynamic range for a linear PCM system using 16-bit
quantizing.

SOLUTION
From Equation (3.11)

DR = 1.76 + 6.02m Db
= 1.76 + 6.02 × 16
= 98.08 dB
X

Bit Rate Increasing the number of bits per sample increases the bit rate, which is
given very approximately by

D = ƒsm (3.12)
where
D = data rate in bits per second
ƒ s = sample rate in samples per second
m = number of bits per sample

Extra bits are often included to detect and correct errors. A few bits,
called framing bits, are also needed to ensure that the transmitter and re-
ceiver agree on which bits constitute one sample. The actual bit rate will
therefore be somewhat higher than calculated above.

EXAMPLE 3.4 Y
Calculate the minimum data rate needed to transmit audio with a sampling
rate of 40 kHz and 14 bits per sample.

SOLUTION
From Equation (3.12)

D = ƒsm
= 40 × 103 × 14
= 560 × 103 b/s
= 560 kb/s
X
94 ! CHAPTER 3

Companding The transmission bandwidth varies directly with the bit rate. In order to
keep the bit rate and thus the required bandwidth low, companding is often
used. This system involves using a compressor amplifier at the transmitter,
with greater gain for low-level than for high-level signals. The compressor
will reduce the quantizing error for small signals. The effect of compression
on the signal can be reversed by using an expander at the receiver, with a
gain characteristic that is the inverse of that at the transmitter.
It is necessary to follow the same standards at both ends of the circuit
so that the dynamics of the output signal are the same as at the input. The sys-
tem used in the North American telephone system uses a characteristic known
as the µ (mu) law, which has the following equation for the compressor:

Vo ln(1 + µv i /Vi )
vo = (3.13)
ln(1 + µ)
where
vo = actual output voltage from the compressor
Vo = maximum output voltage
Vi = the maximum input voltage
vi = the actual input voltage
µ = a parameter that defines the amount of compression
(contemporary systems use µ = 255)
European telephone systems use a similar but not identical scheme
called A-law compression.
Figure 3.6 on page 95 shows the µ-255 curve. The curve is a transfer func-
tion for the compressor, relating input and output levels. It has been normal-
ized, that is, vi/Vi and vo/Vo are plotted, rather than vi and vo.

EXAMPLE 3.5 Y
A signal at the input to a mu-law compressor is positive, with its voltage
one-half the maximum input voltage. What proportion of the maximum
output voltage is produced?

SOLUTION
From Equation (3.13)
Vo ln(1 + µv i /Vi )
vo =
ln(1 + µ)

Vo ln(1 + 255 × 05
. )
=
ln(1 + 255)
= 0876
. Vo
 DIGITAL COMMUNICATION ! 95

This problem can also be solved graphically, using Figure 3.6.

X
FIGURE 3.6
Mu-Law
compression

Digital companding is also possible. The method is to quantize a signal

using a greater number of bits than will be transmitted, and then perform
arithmetic on the samples to reduce the number of bits. This is the way
companding is done in most modern telephone equipment. This type of
companding is part of the coding and decoding process, which is the topic
of the next section.

Coding and The process of converting an analog signal into a PCM signal is called cod-
Decoding ing, and the inverse operation, converting back from digital to analog, is
known as decoding. Both procedures are often accomplished in a single in-
tegrated-circuit device called a codec.
Figure 3.7 is a block diagram showing the steps for converting an analog
signal into a PCM code. The first block is a low-pass filter, required to pre-
vent aliasing. As shown in section 3.2, the filter must block all frequency
components above one-half the sampling rate. This requires a high-order
filter.
96 ! CHAPTER 3

FIGURE 3.7 PCM coding

The next step is to sample the incoming waveform using a sample-

and-hold circuit. There are many such circuits; a simple one is shown in
Figure 3.8. The field-effect transistor Q turns on during the sample time, al-
lowing the capacitor to charge to the amplitude of the incoming signal. The
transistor then turns off, and the capacitor stores the signal value until the
analog-to-digital converter has had time to convert the sample to digital
form. The two operational amplifiers, connected as voltage followers, isolate
the circuit from the other stages. The low output impedance of the first stage
ensures that the capacitor quickly charges or discharges to the value of the
incoming signal when the transistor conducts.
The samples must now be coded as binary numbers. If we are using linear
PCM, all that is required is a standard analog to digital (A/D) converter.
Compression, if required, can be applied to the analog signal, but it is more
common to incorporate the compression into the coding process.
The codecs used in telephony generally accomplish compression by us-
ing a piecewise-linear approximation to the mu-law curve shown earlier in
Figure 3.6. The positive- and negative-going parts of the curve are each di-
vided into seven segments, with an additional segment centered around
zero, resulting in a total of fifteen segments. Figure 3.9 shows the segmented
curve. Segments 0 and 1 have the same slope and do not compress the seg-
ment. For each higher-numbered segment, the step size is double that of
the previous segment. Each segment has sixteen steps. The result is a close
approximation to the actual curve.
The binary number produced by the codec in a telephone system has
eight bits. The first is a sign bit, which is a one for a positive voltage and a
zero for negative. Bits 2, 3, and 4 represent the segment number, from zero to
seven. The last four bits determine the step within the segment. If we nor-
malize the signal, that is, set the maximum input level equal to one volt, the

FIGURE 3.8
Sample-and-
hold circuit
 DIGITAL COMMUNICATION ! 97

FIGURE 3.9
Segmented mu-law
curve

step sizes can easily be calculated as follows: let the step size for the 0 and 1
segments be x mV. Then segment 2 has a step size of 2x, segment 3 a step size
of 3x, and so on. Since each segment has 16 steps, the value of x can be found
as follows.
16(x + x + 2x + 4x + 8x + 16x + 32x + 64x) = 1000 mV
x = 0.488 mV
The relationship between input voltage and segment is shown in Ta-
ble 3.1.

TABLE 3.1 Mu-Law Compressed PCM Coding

Segment Voltage Range (mV) Step Size (mV)

0 0–7.8 0.488

1 7.8–15.6 0.488

2 15.6–31.25 0.9772

3 31.25–62.5 1.953

4 62.5–125 3.906

5 125–250 7.813

6 250–500 15.625

7 500–1000 31.25

EXAMPLE 3.6 Y
Code a positive-going signal with amplitude 30% of the maximum allowed
as a PCM sample.
98 ! CHAPTER 3

SOLUTION
The signal is positive, so the first bit is a one. On the normalized voltage
scale, the amplitude is 300 mV. A glance at Table 3.1 shows that the signal is
in segment 6. That means the next three bits are 110 (6 in binary). This seg-
ment starts at 250 mV and increases 15.625 mV per step. The signal voltage is
50 mV above the lower limit, which translates into 50/15.625 = 3.2 steps.
This is less than halfway from step 3 to step 4, so it will be quantized as
step 3, making the last four bits 0011 (3 in binary). Therefore the code repre-
senting this sample is 11100011.
X

In operation, many modern codecs achieve compression by first encod-

ing the signal using a 12-bit linear PCM code, then converting the 12-bit
linear code into an 8-bit compressed code by discarding some of the bits.
This is a simple example of digital signal processing (DSP). Once an analog
signal has been digitized, it can be manipulated in a great many ways simply
by performing arithmetic with the bits that make up each sample. In the case
of the 12-to-8 bit conversion described here, some precision will be lost for
large-amplitude samples, but the data rate needed to transmit the informa-
tion will be much less than for 12-bit PCM. Since most of the samples in an
audio signal have amplitudes much less than the maximum, there is a gain
in accuracy compared with 8-bit linear PCM.
Briefly, the conversion works as follows. The 12-bit PCM sample begins
with a sign bit, which is retained. The other eleven bits describe the ampli-
tude of the sample, with the most significant bit first. For low-level samples,
the last few bits and the sign bit may be the only non-zero bits. The segment
number for the 8-bit code can be determined by subtracting the number of
leading zeros (not counting the sign bit) in the 12-bit code from 7. The next
four bits after the first 1 are counted as the level number within the segment.
Any remaining bits are discarded.

EXAMPLE 3.7 Y
Convert the 12-bit sample 100110100100 into an 8-bit compressed code.

SOLUTION
Copy the sign bit to the 8-bit code. Next count the leading zeros (2) and sub-
tract from 7 to get 5 (101 in binary). The first four bits of the 8-bit code are
thus 1101. Now copy the next four bits after the first 1 (not counting the sign
 DIGITAL COMMUNICATION ! 99

bit) to the 8-bit code. Thus the next four bits are 1010. Discard the rest. The
corresponding 8-bit code is 11011010.
X
The decoding process is the reverse of coding. It is illustrated in the block
diagram in Figure 3.10. The expansion process follows an algorithm analo-
gous to that used in the compressor. The low-pass filter at the output re-
moves the high-frequency components in the PAM signal that exits from the
digital-to-analog converter.

FIGURE 3.10 PCM decoding

Differential PCM Instead of coding the entire sample amplitude for each sample, it is possible
to code and transmit only the difference between the amplitude of the cur-
rent sample and that of the previous sample. Since successive samples often
have similar amplitudes, it should be possible to use fewer bits to encode the
changes. The most common (and most extreme) example of this process is
delta modulation, which is discussed in the next section.

' 3.4 Delta Modulation

In delta modulation, only one bit is transmitted per sample. That bit is a one
if the current sample is more positive than the previous sample, zero if the
current sample is more negative. Since only a small amount of information
about each sample is transmitted, delta modulation requires a much higher
sampling rate than PCM for equal quality of reproduction. Nyquist did not
say that transmitting samples at twice the maximum signal frequency would
always give undistorted results, only that it could, provided the samples
were transmitted accurately.
Figure 3.11 shows how delta modulation generates errors. In region (i),
the signal is not varying at all; the transmitter can only send ones and zeros,
however, so the output waveform has a triangular shape, producing a noise
signal called granular noise. On the other hand, the signal in region (iii)
changes more rapidly than the system can follow, creating an error in the
output called slope overload.
100 ! CHAPTER 3

FIGURE 3.11 Delta modulation

Adaptive Delta Adaptive delta modulation, in which the step size varies according to previ-
Modulation ous values, is more efficient. Figure 3.12 shows how it works. After a number
of steps in the same direction, the step size increases. A well-designed adap-
tive delta modulation scheme can transmit voice at about half the bit rate of
a PCM system, with equivalent quality.

' 3.5 Data Compression

In the previous section we looked at companding, which we noticed con-
sisted of compression at the transmitter and expansion at the receiver.
Essentially this is an analog technique for improving dynamic range by
increasing the signal-to-noise ratio for low-level signals although, as we
 DIGITAL COMMUNICATION ! 101

FIGURE 3.12 Adaptive delta modulation

saw, it can be implemented using digital signal processing. We now turn

our attention to the bits that result from the analog-to-digital conver-
sion just discussed and consider whether there is any way to reduce the num-
ber of bits that have to be transmitted per second. This reduction is also
called compression, but it is really a completely different process from
the one just described. We shall call it data compression to emphasize the
difference.
Generally, without data compression more bandwidth is required to
transmit an analog signal in digital form. For instance, analog telephony re-
quires less than 4 kHz per channel with single-sideband AM transmission.
Digital telephony conventionally operates at 64 kb/s. The exact bandwidth
requirement for this depends on the modulation scheme but is likely to be
much more than 4 kHz unless the channel has a very high signal-to-noise
radio and an elaborate modulation scheme is used. In order to use digital
102 ! CHAPTER 3

techniques in wireless communication, it is very desirable to reduce the

bandwidth to no more, and preferably less, than that needed for analog
transmission.

Lossy and There are two main categories of data compression. Lossless compression in-
Lossless volves transmitting all of the data in the original signal but using fewer bits.
Compression Lossy compression, on the other hand, allows for some reduction in the quality
of the transmitted signal. Obviously there has to be some limit on the loss in
quality, depending on the application. For instance, up until now the expecta-
tion of voice quality has been less for a mobile telephone than for a wireline
telephone. This expectation is now changing as wireless telephones become
more common. People are no longer impressed with the fact that wireless tele-
phony works at all; they want it to work as well as a fixed telephone.
Lossless compression schemes generally look for redundancies in the
data. For instance, a string of zeros can be replaced with a code that tells the
receiver the length of the string. This technique is called run-length encod-
ing. It is very useful in some applications: facsimile (fax) transmission, for
instance, where it is unnecessary to transmit as much data for white space on
the paper as for the message.
In voice transmission it is possible to greatly reduce the bit rate, or even
stop transmitting altogether, during time periods in which there is no
speech. For example, during a typical conversation each person generally
talks for less than half the time. Taking advantage of this to increase the
bandwidth for transmission in real time requires there to be more than one
signal multiplexed. When the technique is applied to a radio system, it also
allows battery-powered transmitters to conserve power by shutting off or
reducing power during pauses in speech.
Lossy compression can involve reducing the number of bits per sample
or reducing the sampling rate. As we have seen, the first reduces the
signal-to-noise ratio and the second limits the high-frequency response of
the signal, so there are limits to both methods. Other lossy compression
methods rely on knowledge of the type of signal, and often, on knowledge of
human perception. This means that voice, music, and video signals would
have to be treated differently. These more advanced methods often involve
the need for quite extensive digital signal processing. Because of this, they
have only recently become practical for real-time use with portable equip-
ment. A couple of brief examples will show the sort of thing that is possible.

Vocoders A vocoder (voice coder) is an example of lossy compression applied to human

speech. A typical vocoder tries to reduce the amount of data that needs to be
transmitted by constructing a model for the human vocal system. Human
sounds are produced by emitting air from the lungs at an adjustable rate. For
 DIGITAL COMMUNICATION ! 103

voiced sounds this air causes the vocal cords to vibrate at an adjustable fre-
quency; for unvoiced sounds the air passes the vocal cords without vibrating
them. In either case, the sound passes through the larynx and mouth, which
act as filters, changing the frequency response of the system at frequent in-
tervals. Typically there are from three to six resonant peaks in the frequency
response of the vocal tract.
Vocoders can imitate the human voice with an electronic system. Mod-
ern vocoders start with the vocal-tract model above. There is an excitation
function, followed by a multi-pole bandpass filter. Parameters for the excita-
tion and the filter response must be transmitted at intervals of about 20 ms,
depending on the system. Vocoders of this type are known as linear predictive
coders because of the mathematical process used to generate the filter param-
eters from an analysis of the voice signal.
The first step in transmitting a signal using a vocoder is to digitize it in
the usual way, using PCM, generally at 64 kb/s. Then the signal is analyzed
and the necessary excitation and filter parameters extracted. Only these pa-
rameters need to be sent to the receiver where the signal is reconstructed.
The transmitted data rate is typically in the range of about 2.4 to 9.6 kb/s,
allowing a much smaller transmission bandwidth than would be required
for the original 64 kb/s rate.
There are two main ways of generating the excitation signal in a linear
predictive vocoder. In pulse excited linear predictive (PELP or sometimes
RPELP, for regular pulse excited linear predictive) vocoders, a white noise gener-
ator is used for unvoiced sounds, and a variable-frequency pulse generator
produces the voiced sounds. The pulse generator creates a tone rich in har-
monics, as is the sound produced by human vocal cords. Both sources have
variable amplitudes. Figure 3.13 illustrates the process at the receiver.
Residual excited linear predictive (RELP) vocoders, on the other hand,
apply the inverse of the filter that will be used at the receiver to the voice
signal. The output of this filter is a signal that, when applied to the receiver
filter, will reproduce the original signal exactly. Figure 3.14 shows how this

FIGURE 3.13 PELP vocoder

104 ! CHAPTER 3

FIGURE 3.14 Generation of excitation signal using codebook

process works at the transmitter. The residual signal is too complex to trans-
mit exactly with the available bit rate, so it must be represented in a more
economical way. One method is to compare it with values in a table, called a
codebook, and transmit the number of the closest codebook entry. The re-
ceiver looks up the codebook entry, generates the corresponding signal, and
uses it instead of the pulse and noise generators shown in Figure 3.13. Many
other vocoder variations are possible as well.
Reasonable quality can be achieved with vocoders using data rates much
lower than those required for PCM. So far, the quality is not quite as good as
for straightforward PCM, however.
It should be obvious that vocoders are intended for use with voice only;
whereas, the PCM system described above can be used to send any 64 kb/s
data stream, including music, fax, or computer files. None of these will work
properly with a vocoder. Vocoders even tend to give a somewhat unnatural
quality to human speech. Still, the gain in bit rate and hence bandwidth,
compared to PCM, is so great that vocoders are very common in digital wire-
less voice communication.

' Summary The main points to remember from this chapter are:
( Modern communication systems are often a mixture of analog and digital
sources and transmission techniques. The trend is toward digital systems.
( Modern digital systems have better performance and use less bandwidth
than equivalent analog systems.
( An analog signal that is to be transmitted digitally must be sampled at
least twice per cycle of its highest-frequency component. Failure to do so
creates undesirable aliasing.
( PCM requires that the amplitude of each sample of a signal be converted
to a binary number. The more bits used for the number, the greater the ac-
curacy, but the greater the bit rate required.
( Delta modulation transmits only one bit per sample, indicating whether
the signal level is increasing or decreasing, but it needs a higher sampling
rate than PCM for equivalent results.
 DIGITAL COMMUNICATION ! 105

( The signal-to-noise ratio for either PCM or delta modulation signals can
often be improved by using companding.
( Lossless compression eliminates redundant data bits, thereby reducing
the bit rate with no effect on signal quality.
( Lossy compression compromises signal quality in order to reduce the bit
rate. For voice transmissions, vocoders are often used to achieve great re-
ductions in bit rate.

( Equation List
fa = fs − fb (3.8)

Eb E sin πτ / T
v (t ) = sin ω bt + b sin(ω s + ω b )t (3.9)
T πτ

E b sin πτ / T
− sin(ω s − ω b )t
πτ

N = 2m (3.10)

DR = 1.76 + 6.02m dB (3.11)

D = ƒsm (3.12)

Vo ln(1 + µv i /Vi )
vo = (3.13)
ln(1 + µ)

( Key Terms
aliasing distortion created by using too low a sampling rate when coding
an analog signal for digital transmission
codec device that converts sampled analog signal to and from its PCM or
delta modulation equivalent
coding conversion of a sampled analog signal into a PCM or delta
modulation bitstream
companding combination of compression at the transmitter and
expansion at the receiver of a communication system
106 ! CHAPTER 3

decoding conversion of a PCM or delta modulation bitstream to analog

samples
delta modulation a coding scheme that records the change in signal
level since the previous sample
digital signal processing (DSP) filtering of signals by converting them to
digital form, performing arithmetic operations on the data bits, then
converting back to analog form
flat-topped sampling sampling of an analog signal using a sample-and-
hold circuit, such that the sample has the same amplitude for its
whole duration
foldover distortion see aliasing
framing bits bits added to a digital signal to help the receiver to detect
the beginning and end of data frames
natural sampling sampling of an analog signal so that the sample
amplitude follows that of the original signal for the duration of the
sample
pulse-amplitude modulation (PAM) a series of pulses in which the
amplitude of each pulse represents the amplitude of the information
signal at a given time
pulse-code modulation (PCM) a series of pulses in which the amplitude
of the information signal at a given time is coded as a binary
number
pulse-duration modulation (PDM) a series of pulses in which the
duration of each pulse represents the amplitude of the information
signal at a given time
pulse-position modulation (PPM) a series of pulses in which the timing
of each pulse represents the amplitude of the information signal at a
given time
pulse-width modulation (PWM) see pulse-duration modulation (PDM)
quantizing representation of a continuously varying quantity as one of a
number of discrete values
quantizing errors inaccuracies caused by the representation of a
continuously varying quantity as one of a number of discrete values
quantizing noise see quantizing errors
run-length encoding method of data compression by encoding the
length of a string of ones or zeros instead of transmitting all the one
or zero bits individually
 DIGITAL COMMUNICATION ! 107

slope overload in delta modulation, an error condition that occurs when

the analog signal to be digitized varies too quickly for the system to
follow
vocoder circuit for digitizing voice at a low data rate by using knowledge
of the way in which voice sounds are produced

( Questions
1. Give four advantages and one disadvantage of using digital (rather than
analog) techniques for the transmission of voice signals.
2. Explain the necessity for sampling an analog signal before transmitting
it digitally.
3. What is the Nyquist rate? What happens when a signal is sampled at less
than the Nyquist rate?
4. Explain the difference between natural and flat-topped sampling.
5. (a) List three types of analog pulse modulation.
(b) Which pulse modulation scheme is used as an intermediate step in
the creation of PCM?
(c) Which pulse modulation scheme also finds use in audio amplifiers
and motor speed-control systems?
6. What is meant by the term quantizing noise?
7. For a PCM signal, describe the effects of:
(a) increasing the sampling rate
(b) increasing the number of bits per sample
8. (a) Briefly explain what is meant by companding.
(b) What advantage does companded PCM have over linear PCM for
voice communication?
9. How does differential PCM differ from standard PCM?
10. Explain why the sampling rate must be greater for delta modulation
than for PCM.
11. What is meant by slope overload in a delta modulation system? How
can this problem be reduced?
12. What are the two functions of a codec? Where in a telephone system is
it usually located?
13. Explain briefly how µ-law compression is implemented in a typical
codec.
108 ! CHAPTER 3

14. Explain the difference between lossless and lossy data compression.
Give an example of each.
15. How do vocoders model the human vocal cords? How do they model
the mouth and larynx?
16. What gives vocoders their somewhat artificial voice quality?
17. Does digital audio always have higher quality than analog audio?
Explain.

( Problems
1. It is necessary to transmit the human voice using a frequency range
from 300 Hz to 3.5 kHz using a digital system.
(a) What is the minimum required sampling rate, according to theory?
(b) Why would a practical system need a higher rate than the one you
calculated in part (a)?
2. The human voice actually has a spectrum that extends to much higher
frequencies than are necessary for communication. Suppose a fre-
quency of 5 kHz was present in a sampler that sampled at 8 kHz.
(a) What would happen?
(b) How can the problem described in part (a) be prevented?
3. A 1-kHz sine wave with a peak value of 1 volt and no dc offset is sampled
every 250 microseconds. Assume the first sample is taken as the voltage
crosses zero in the upward direction. Sketch the results over 1 ms using:
(a) PAM with all pulses in the positive direction
(b) PDM
(c) PPM
4. The compact disc system of digital audio uses two channels with TDM.
Each channel is sampled at 44.1 kHz and coded using linear PCM with
sixteen bits per sample. Find:
(a) the maximum audio frequency that can be recorded (assuming
ideal filters)
(b) the maximum dynamic range in decibels
(c) the bit rate, ignoring error correction and framing bits
(d) the number of quantizing levels
5. Suppose an input signal to a µ-law compressor has a positive voltage
and an amplitude 25% of the maximum possible. Calculate the output
voltage as a percentage of the maximum output.
 DIGITAL COMMUNICATION ! 109

6. Suppose a composite video signal with a baseband frequency range

from dc to 4 MHz is transmitted by linear PCM, using eight bits per sam-
ple and a sampling rate of 10 MHz.
(a) How many quantization levels are there?
(b) Calculate the bit rate, ignoring overhead.
(c) What would be the maximum signal-to-noise ratio, in decibels?
(d) What type of noise determines the answer to part (c)?
7. How would a signal with 50% of the maximum input voltage be coded
in 8-bit PCM, using digital compression?
8. Convert a sample coded (using mu-law compression) as 11001100 to a
voltage with the maximum sample voltage normalized as 1 V.
9. Convert the 12-bit PCM sample 110011001100 to an 8-bit compressed
sample.
10. A typical PCS system using a vocoder operates at 9600 b/s. By what fac-
tor has the amount of data required been reduced, compared with stan-
dard digital telephony?