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Heriot-Watt University
Edinburgh Electrical, Electronic & Computer Engineering

Digital Communications 1 –Revision of the fundamentals

1 Objectives
After this unit, you should be able to:
Define signal bandwidth
Distinguish between high- low- and band-pass responses
List the baseband frequencies for a range of sources
List the transmission frequencies for a range of physical media
Convert power gains and signal-to-noise ratios into dB
1.1 Reading list
You will find this material in any elementary comms textbook, especially the following
Usher and Guy, "Information and Communication for Engineers"
Bateman, A, " Digital Communications"
Glover and Grant, "Digital Communications"

1.2 Communication processes

Any communication channel transfers information from a source to a destination: (Fig 1)

Information
Source ⇒ Communication
Channel ⇒ Information
Destination

The channel contains all of the signal processing, as well as the actual communication medium, such as the
cable or radio link.
• The signal processing involves some or all of the following:-
Generation by the source of information (How do we measure information?)
Sampling to digitise analog signals (What sampling rate should we use?)
Coding & to suit the medium (cables and radio require different transmission codes)
decoding data compression (MP3, JPEG, WINZIP – reduce transmission times
or storage)
error correction (all signals are corrupted by noise and interference)
Modulation & Analog (AM and FM, V90 or 56kbit/s modems)
Demodulation Digital (TV, radio, mobile phones – GSM/2.5G/3G, broadband modems)
Frequency for efficient transmission and multiplexing (MW/VHF/UHF broadcasting:
conversion µwaves for mobile phones and satellites:
fibreoptic or coaxial cable)
and the usual processes of Amplification and Filtering
• Some common information sources are:
Audio - Speech, Music (Radio, TV, Mobile phones, DVD, CD)
Visual - Still/Moving images (TV, VCR, Mobile phones, DVD)
Computer Data File transfer/storage, internet up/downloads
Facsimile (Fax) Documents
Telemetry Measurements, Radar & Sonar systems

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 1.3 The specification of signals
An electrical signal can be specified in terms of either waveform or spectrum
• Waveform: analogue/digital, periodic/aperiodic,
• Spectrum: bandwidth, signal-to-noise power ratio.1

The bandwidth of a signal is the difference between the maximum and minimum
frequencies of significant amplitude in that signal.

• How to define “significant amplitude”:

3-dB bandwidth2, or “half-power” bandwidth The channel must have an adequate
frequency response to carry the signal or
60-dB bandwidth,
else it may be distorted and lose
noise bandwidth information
• Classification of frequency response: Lowpass: bandpass: highpass (Fig 2).

amplitude
amplitude (a) Waveform

(b) Lowpass spectrum

time

frequency

amplitude amplitude
(c) Bandpass spectrum

(d) Highpass spectrum

frequency frequency

Fig 2 Descriptions of signal bandwidth and system frequency response.

1.4 Baseband and Bandpass signals

The baseband is defined as the spectrum of frequencies produced by the signal source before it is
processed for transmission by coding, modulation or frequency conversion3. It will be either low-pass or band-
pass, never high-pass. If it is low-pass it will extend down to 0 Hz and will have a dc component.

Speech 300 Hz ⇒ 3.4 kHz (Telephone quality)

HiFi Audio 50 Hz ⇒ 15 kHz
Video 0 ⇒ 6 MHz
Data Typically 1 Hz per symbol/s
2400 symbols/s for telephone links
106 symbols/s for Data Networks

Table 1 Typical Baseband frequencies

1 S/N ratio or "SNR"

2 see appendix 1.1
3 Also known as frequency shifting, frequency changing, mixing or heterodyning.

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1.4.1 The bandpass signal
For electromagnetic transmission - radio and television broadcasting for example - the usable electromagnetic
spectrum is divided, by International Agreement, into channels of varying bandwidth to suit the nature of the
information. Frequency shifting or the process of modulation moves the baseband up to the transmission or
"carrier" frequency of the allocated channel, or to otherwise suit the physical medium, and produces a
“bandpass” signal.

Copper cable 0 ⇒ ~100 MHz AM/FM Radio 150 kHz ⇒ 100 MHz

Terrestrial TV 450 MHz ⇒ 850 MHz Mobile phones 900 MHz ⇒ 1800MHz

Satellite radio/TV 11 GHz ⇒ 12 GHz Fibre-optic cable ~3 x 1014 Hz (1 µm)

Table 2Transmission frequencies

1.5 Signal power measurements in communication systems
The power of a signal can be calculated from
P = V2/R 1
I
Where V is the rms voltage, (root mean square)and
V R
T ∫T
V =12 2
v ( t )dt 2

calculates the mean-square voltage from the waveform4.

For a system with input power Pi and output power Po,
the Power gain, g = Po/Pi 3 Fig 3 Definition of power
If the impedances are the same at input and output this reduces to
g = Vo2/Vi2 4
2
where V is the mean-square voltage.
•
2
It has become customary to refer to V as the signal power (it would only be the true power if R =
1Ω), and to avoid confusion we will use
S = V2 5
A gain is dimensionless: a numerical ratio with no units. The ratio is normally converted into decibels5 so that
G = 10 log10 (So/Si) dB 6
• The use of decibels are so convenient and widespread that it is safe to assume that every
measurement or calculation of gain is carried out in dB, unless otherwise indicated.

1.6 Signal-to-noise ratio

S
The quality of a communication channel is largely determined by the Signal to Noise ratio /N.
This can be either the power ratio, or the mean-square voltage ratio (see above for the difference), and is
normally expressed in dB. Thus

SNR = 10 log10 (S/N) dB 7

• For Hi-Fi reproduction of speech or music we would expect a signal-to-noise ratio of 60 dB or

more
• Data channels can give a low error rate with a SNR less than 0 dB.

4The interval T is either the period, for a periodic signal, or sufficiently long to capture a typical sample of a
non-periodic signal
5 The "bel" is named after Alexander Graham Bell, but is rather a large unit, hence the use of decibels, or dB

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1.7 SNR, Bandwidth and channel capacity
We would naturally expect the capacity of a communication channel to depend on
• The Signal-to-Noise ratio
• The bandwidth of the channel
This is expressed in the Hartley-Shannon Law:
C = Blog2(1 + S/N) 8
• B is the channel bandwidth in hertz
S
• /N is the numerical signal-to-noise power ratio, not in dB
• The logarithm is to base 2
• the capacity C is in bit/s and is the maximum theoretical capacity of a channel

1.8 The electromagnetic spectrum

It is convenient to divide the spectrum into decades, beginning at 30 kHz. Below this frequency, it is not really
usable except for em communication under water6, which rapidly attenuates higher frequencies, and such low
frequencies only permit a very low bandwidth signal to be used.

Frequency Band Wavelength Typical application

30 – 300 kHz 10 km – 1 km LW broadcasting Radio beacons

300 – 3000 kHz 1000 m – 100 m MW broadcasting Maritime radio

3 MHz –30 MHz 100 m – 10 m SW Broadcasting

30 MHz – 300 MHz 10 m – 1 m FM broadcasting Aircraft radio Radio-

telephones
0.3 – 3 GHz 1 m – 10 cm TV, Radar, Mobile phones Microwave links,
Microwave ovens
3 GHz – 30 GHz 10 cm – 1 cm Satellite broadcasting, Microwave links
Radar

You will remember that frequency and wavelength are related by the speed of propagation, and
fλ = c, where c (the speed of light) is 3 x 108 m/s in free space.

6 Not to be confused with Sonar signalling, which uses sound-pressure , not em waves, and can operate at MHz
frequencies

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Appendix 1.1 Why we use decibels
As we have observed, the ear responds to sound pressure with a non-linear function (Fig A1.1)
It is reasonable to assume that there will be some threshold
pressure PT below which no sound is perceived, and we can
loudnesss
then write the loudness
L = k log10P/PT

sound pressure
PT If the pressure scale is replaced with
x = log10P/PT
loudnesss
then L = kx is a linear function and the x-axis has a
logarithmic scale where equal increments in x give equal
increments in loudness but correspond to a 10x increase in
pressure.

0 1 2 3 x
1 10 100 1000 P/P T Fig A1.1

There are other good practical reasons why we use dB. In most systems signals are processed in a chain or
cascade of sub-systems: (Fig A1.2):

P1 P2 P3
g1 g2 g3

Fig A1.2 Cascaded gain

If g is the power gain of a sub-system, expressed as the ratio output/input, then
P2/P1 = g1
P3/P2 = g2
and P3/P1 = g1g2 etc
and we see that the gain ratios are multiplied together as the signal passes down the system.
If we take the logarithm of the gain - it does not actually matter what base we use, although base-10 is normal,
then
log10 P3/P1 = log10 g1g2
= log10 g1 + log10 g2
and multiplication is replaced by addition, which is good for engineers who find it easier to add than multiply.
• Hence G = log10 g defines the gain in bels, and G = G1 + G2 +….
Noting that g = 10G from the definition of a logarithm, we see that 1 bel represents a gain ratio of 101 or 10
times. For reasons which are lost in the midst ot time it was decided that this was too large, and the practical
log-gain unit is the decibel.
• Hence G = 10log10 g defines the gain in decibels.

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If we consider the voltage gain, then P1 = V12/R1 etc.
For convenience we will take all the input/output impedances to be equal, a condition known as matchingA1,
and then
G = 10log10 V22/V12 = 20log10 V2/V1
Thus

G = 20log10 V2/V1 dB or G = 10log10 P2/P1 dB

and V2/V1 = 10G/20 or P2/P1 = 10G/10

Examples:
A power gain of 2 is 10log10 2 = 3.010 dB, which is usually approximated by "3 dB". The corresponding
voltage gain-ratio is of course √2, and 103/20 = 1.41 ≈ √2.
A power gain of 0.5 is 10log10 0.5 = -3 dB, so the "half-power point" of a circuit is also the -3 dB point. (And
the voltage gain is 1/√2 = 0.707).
A power gain of 10 is, as we have already noted, also 10 dB - very confusing! But of course a voltage gain of
10 is 20 dB. (A gain of 10 dB is a voltage ratio of 3.16 - work it out!).

Q. Does it work if there is also a phase shift?

A. Yes: look at this:
If each stage has a phase shift, then we can use a complex gain g 1e jθ1 , etc giving
V3 jθ 2
= g 1e jθ1 g 2 e
V1
= g 1 g 2 e j( θ1 + θ2 )
Now, we said that the base of the logarithm need not be 10: so consider natural logs:
V3
ln = ln g 1 g 2 + j( θ 1 + θ 2 )
V1
= (G 1 + G 2 ) + j(θ 1 + θ 2 )

So we add the log-gains as before, and add the phase shifts as normal.
Q. But these are not decibels…
A Quite right, because we used natural logs, but remember how to change the base?
If y = log10 x then x = 10y
Now put 10 = ea, which means that a = ln 10
Thus x = (ea)y and therefore ay = ln x
Or y = (ln x)/a
ie log10 x = (ln x)/(ln 10)
= 0.434 ln x
The natural-log gain unit is the neperA2, so 1 neper = 0.434 dB (or 1 dB = 2.302 nepers). Nepers are actually
used by transmission-line engineers.
Q So what do I actually do?
A Just add the gains in dB, and the phases in degrees and stop worrying.

A1 Even if the impedances are not equal, we usually ignore the fact and apply this simple formula - it makes no
difference if all that we are interested in is a change in gain rather than its absolute value.
A2 Apparently named after Napier, who devised natural logarithms. (So why not napers?)

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Q You said that there were several reasons for using dB (or nepers!)
A Here is another reason. Remember that the frequency response of an RC low-pass filter is given by
Vo
= 1
Vi 1 + jωCR
= 1 〈− tan −1 ωCR
1 + (ωCR ) 2

Considering the amplitude response in dB,

V2
= −10 log 10 {1 + (ωCR ) 2 }
V1 dB

Now if ωCR << 1, G ≈ -10 log101 = 0 dB

And if ωCR >> 1, G ≈ -20 log10 ωCR dB, which on a logarithmic frequency scale gives a straight line of slope
-20 dB/decade, - and engineers prefer to draw straight lines….
At ωCR = 1, G = -10 log10 2, which is, of course -3 dB(!) which gives the "half-power point" at a frequency of
1/CR rad/s.

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