EM 437

COMMUNICATION SYSTEMS II

LABORATORY MANUAL

2006-2007 FALL

TABLE OF CONTENTS
page
LABORATORY RULES...................................................................................................................3
CHAPTER 1: PULSE CODE MODULATION ..............................................................................4
1. 1. Introduction ................................................................................................................................4
1. 2. The Apparatus ............................................................................................................................4
1. 3. The Modulation Process ............................................................................................................4
1. 4. Quantising Noise.........................................................................................................................6
1. 5. References ...................................................................................................................................8
CHAPTER 2: SAMPLING AND TIME DIVISION MULTIPLEX .............................................9
2. 1. Introduction ................................................................................................................................9
2. 2. The Apparatus ............................................................................................................................9
2. 3. The Sampling Process-Theory ..................................................................................................9
2. 4. Time Division Multiplexing .....................................................................................................14
2. 5. Observation of Signals .............................................................................................................14
2. 6. References .................................................................................................................................15
Appendix 1. Convolution .................................................................................................................16
EXPERIMENT 1 : PCM CODING.............................................................................................17
EXPERIMENT 2 : PCM QUANTISING LEVELS...................................................................20
EXPERIMENT 3 : PCM QUANTISING NOISE ......................................................................22
EXPERIMENT 4: SAMPLING AND TIME DIVISION MULTIPLEX....................................26

Gazi University
Department of Electrical and Electronics Engineering
EM437 Communication Systems II Lab.

LABORATORY RULES

Everyone should attend experiments on the determined days and hours.

If one does not attend more than one experiment he/she will take the grade DY.
There will be four experiments.
Reports should include:
Name and number of writer
Name, purpose, theory and the procedure of experiment
Comments of writer
There will be a quiz for every experiment.
Grading will be as follows:
Experiments
: 25%
Quiz
: 25%
Performance
: 10%
Final Examination : 40%

CHAPTER 1: PULSE CODE MODULATION

1. 1. Introduction
Pulse Code Modulation(PCM) is by far the most important digital modulation method. It is
used extensively in data and communication channels and is widely used for transmission of telephony
between exchanges. In fact the words analogue-to-digital conversion (A/D) often imply PCM
although other A/D methods such as delta modulation are possible.
In PCM the analogue signal is first sampled and the amplitude of each sample is used to
encode a stream of pulses. The coding used is often binary because binary electronic devices are
cheap.
1. 2. The Apparatus
Fig.1.1 shows the apparatus, which comprises a pulse code modulator and demodulator. The
input to the modulator can have a signal of any waveform or amplitude applied to it, and a separate
d.c. level control is supplied on the equipment. A choice of 3,4 or 8 bit encoding is available at the
modulator. The signal can be reconstituted by the pulse code demodulator. A difference amplifier is
also supplied to allow the signal into the p.c.m modulator to be compared with the signal out of the
p.c.m demodulator in order to establish the loss of information(i.e. added noise) as a function of the
encoding, frequency and amplitude of the input signal.
The following additional apparatus is required:
(a) Audio Signal Generator
(b) Oscilloscope
1. 3. The Modulation Process
The signal to be transmitted is first sampled. As demonstrated in the experiment
E15g(Sampling and Time Division Multiplex), the sampling rate should be at least twice the highest
frequency in the waveform, otherwise aliasing distortion will occur.
Each sample is then encoded into an m-bit binary word. The choice of the number of bits per
word(m) is rather important and as usual there is a conflict between opposing considerations. On the
one hand m bits means that there can only be 2m possible codes to described the sample amplitude.
On the other hand large values of a m mean that a higher bit rate is needed to transmit the required
number of samples per second. Thus the system design requires that the minimum value of m be
chosen which is consistent with adequate the signal with an m bit code.
If m is chosen to be 3 then there 8 possible amplitude levels, thus,
Code
111
110
101
100
011
010
001
000

Level
7
6
5
4
3
2
1
0

Fig. 1.1

A 3-bit code means therefore that signal only be described by 8 voltages.

This may be demonstrated on the apparatus using the d.c. supply provided. This should be
connected to the signal input and the code at the modulator output can be observed on the
oscilloscope. It is necessary to identify the beginning of a code word and this is achieved by
simultaneously displaying the word pulse. The decoded d.c voltage levels can be observed at the
demodulator output after connecting the demodulator to the modulator. In this way the voltage levels
for each of the 8 possibilities can be established.
The experiment should be repeated for a 4-bit code with its 16 possibilities, and an 8 bit code
with its 256 possibilities.
1. 4. Quantising Noise
When a 3-bit word is being used there are only 8 possible amplitudes that can be transmitted
and there is therefore an error between the actual level of the signal and the transmitted signal. The
process of making the amplitudes of the samples correspond to discrete levels is termed quantising and
the amplitude errors introduced constitute quantising noise. Fig.1.2 shows an analogue signal and its
quantised version. The error signal, also shown, is the quantised signal subtracted from the analogue
signal.
If the voltage difference between the quantising levels is s then the error signal is a sawtoothlike waveform of peak-to-peak amplitude s. the mean square value of such a waveform is s2/12 and so
this quantifies the noise in PCM systems. The fact that this noise waveform contains very high
frequencies does not affect the amount of noise falling into the signal board because of the sampling
process.
Now we have to decide what the value of s is. Suppose that level 0 in a 3-bit system is Vc
volts and level 7 is +Vc volts then the value of s is 2 Vc/7, or in general
s= 2 Vc / (2m-1)

. (1.1)
Signal

Quantised signal
S

S
Error signal

Fig. 1.2

The quantising noise NQ is thus given by :

NQ =

2
2
s 4V c
12 12 x 22 m

.. (1.2)

+Vc

-Vc

The signal to noise ratio obviously depends also on the amplitude of the signal. Fig 1.3 shows
the quantising levels for a 3-bit code with waveforms of 3 different amplitudes. Waveform A does not
use the whole of the amplitudes available and is, therefore, unnecessarily small. Waveform C is too
large because it spends a significant time out of the range of the quantising levels and therefore
becomes severely clipped. Waveform B is just right. For a sine wave, therefore, it is easy to decide
that the amplitude should be Vc. the mean square value (s) of the sine wave is Vc/2 and so the signal to
quantising noise ratio becomes:

NQ

3x 22m
2

(1.3)

The signal to noise ratio for various values of m are shown as follows (note the approximation
in equation 1.2 causes errors for low values of m):
m

S/NQ (dB)

3
4
5
6
7
8

9.8
16.2
31.9
37.9
43.9
49.9

For signals with noise-like characteristics the problem is more difficult because it is not so
easy to decide what the optimum amplitude should be. For speech signals there is the additional
problem that some people talk loudly and others talk more quietly. Often the quantising grids are
unevenly spaced giving a process called companding and this more nearly matches the quantising grid
to the statistics of the signal.
The quantising of a sinusoidal signal can be observed by applying a signal of 100 Hz to the
signal input and adjusting the oscilloscope for stable triggering. The modulator output can be seen to
be cyclically reading through the code alphabet. By connecting the modulator output to the
demodulator input and observing the demodulator output, the quantising effect of the pulse code
modulation process can be clearly seen.
The amplitude of the signal should now be increased so that it exceeds the extreme levels as
determined by the d.c. test. Clipping of the sinusoidal signal can now be observed, high amplitudes
leading to longer times during which the sinusoid is clipped.
Applying a sinusoidal signal to both the signal input to terminal B of the difference amplifier,
and connecting the demodulator output to terminal A, allows the quantising noise to be examined. The
observed shapes should be explained as the amplitude, bits per word, bit rate and input signal
frequency are varied. If an r.m.s. meter is available it can be used to measure the quantising noise
power and the signal input power. This measurement can be made for various input amplitudes and a
graph plotted of signal to noise ratio versus input amplitude. As explained above this should peak
when the signal amplitude is near Vc and the signal to noise ratio is then given by equation 1.3.
1. 5. References
1. P.B. Johns T.R. Rowbotham, Communications Systems Analysis, Butterworth 1972.
2. J.A. Betts, Signal Processing, Modulation and Noise, English Universities 1970.

CHAPTER 2: SAMPLING AND TIME DIVISION MULTIPLEX

2. 1. Introduction
In all telecommunications networks there is a need to interconnect switching centers and
telephone exchanges as economically as possible. Usually the volume of traffic over these routes
makes it attractive to transmit as much information as possible over each cable. Clearly if the distance
between transmitter and receiver is short the cost of the terminal equipment which combines
information channels may exceed the cost of the installing extra cable pairs, thus there are certain
distances below which limits can be set on the number of telephone channels it is worthwhile to
combine for the purposes of transmission. Although it might appear that the economist could draw a
continuous curve relating distance to the number of telephone channels, it must also be borne in mind
that it is more economic to develop and manufacture a small range of products rather than a large,
flexible range. Channel combining (multiplexing) equipment operating in the frequency domain is
known as frequency division multiplexing(F.D.M) while in the time domain it is known as time
division multiplexing(T.D.M). The quantum of telephone channels in F.D.M. is the group, which
comprises twelve telephone channels, and that for T.D.M. is either twenty-four or thirty telephone
channels. The common European standard has now been agreed as thirty channels, which corresponds
to 2048 kbit/s. Further agreed orders in the hierarchy are at information rates of 8448 kbit/s, 34368
kbit per second and 139264 kbit per second. The basic processes in building this hierarchy from an
analog telephone channel occupying 300Hz to 3400Hz are sampling at 8kHz, 8-bit encoding(as
described in the pulse code modulation experiment), and time division multiplexing of 30 of these 64
kbit/s channels. This experiment investigates the first and last of these processes.
2. 2. The Apparatus
The apparatus is shown in fig.2.1. It comprises a sampling source, which may varied in
frequency or sample pulse width, a multiplexer and a demultiplexer. The multiplexer accepts two or
four input channels, samples each, and interleaves the samples. The signal on one of these channels is
a waveform containing approximately the first and third harmonics of a 1kHz signal. The output from
the multiplexer may be observed, or may be transmitted into the demultiplexer which separates the
two channels, and passes the pulse train of each through a low pass filter to reconstitute the original
signals.
An oscilloscope and an audio signal generator are required in addition to the apparatus.
2. 3. The Sampling Process-Theory
In certain communication processes, such as pulse code modulation system described in
Chapter 1, it is necessary to sample a waveform at regular intervals in order to communicate discrete
information rather than continuous information. The process of sampling is equivalent to multiplying
the waveform to be sampled by a series of regularly spaced delta functions as shown in fig.2.2.
Such a series of delta pulses is termed the sampling function which has the interesting
property that an infinite series of delta pulses in the time domain has a spectrum which is also an
infinite delta series in the frequency domain. Communications engineers often have to work
simultaneously in both the frequency and the time domain, and probably the best known rule which
connects manipulations in these two domains is that a multiplication of waveforms in the time
domain transforms in the convolution of their corresponding amplitude spectra in the frequency
domain. Convolution may sound as if its a difficult process, and indeed may be so mathematically,
but it is in fact a simple geometrical process which is described in more detail in the appendix to this
chapter. Thus if sampling the multiplication of the analog waveform by a delta series in the time
domain, the spectrum of the sample signal is the convolution of the analog waveform spectrum with
another delta series. This is shown in fig.2.3.

Fig.2.1

10

t
(a)Waveform

t
(b) Sampling function

t
(c)=(a)x(b)

Fig. 2.2

11

-fm

fm

1/T

f
fm

Fig. 2.3
If T is the interval between pulses in the time domain, i.e. in fig.2.2, then the corresponding
interval between the frequencies which contain signal energy is 1/T. Consider an analog waveform
which has a spectrum which extends from zero Hz to an upper limit of f m Hz. It can now be seen that
provided 1/T is greater than 2 f m then a complete replica of the spectrum of the sampled signal lies
below the frequency 1/2T and the introduction of a low pass filter would restore the original signal
unchanged. If, however, the frequency 1/T is less than 2 f m then overlap of the spectra of the sampled
signal will occur resulting in distortion. This mechanism of distortion is sometimes referred to as
aliasing and is shown in fig.2.4.
The preceding argument holds true even if the analog signal spectrum begins at a frequency
above zero Hz, and indeed in the limit it could be a sinusoid with energy only at frequency f m . In
general, therefore, if a waveform has frequencies in its spectrum extending from a lower frequency
limit to an upper frequency limit f m Hz it is possible to convey all the information in that waveform
by 2 f m or more equally spaced samples per second of the amplitude of the waveform. This rate is
often referred to as the Nyquist sampling rate.

12

-fm

fm

f
1/T

fm
Fig.2.4

In practice, an analog signal is sampled with pulses which have a nonzero width. The way in
which the pulse width affects the spectrum will be used to demonstrate the effect in practice. Thinking
geometrically again, a series of a broad samples is a delta series convolved with a single broad pulse.
Using x for the multiplication process and * for the convolution process, fig.2.5(a) shows the
procedure in constructing the practical sampled waveform. Using the rule referred to before where x
and * change places when moving between time and frequency domains, fig. 2.5(b) shows how the
practical spectrum is constructed. The extra ingredient used in moving from fig. 2.5(a) to fig.2.5(b) is
the relationship between a square pulse and its amplitude density spectrum. This is shown in fig.2.6
which also shows that the narrower the pulse the broader the amplitude density spectrum between its
central zero crossing points.
In the limit of a zero width(delta) pulse this amplitude density spectrum is flat, which, when
multiplied by any other spectrum, does not alter its shape, as is excepted.

(a)

(b)
f

x
f

Fig.2.5

13

t
=

=
f

1/

Fig. 2.6
2. 4. Time Division Multiplexing
Time division multiplexing is the process whereby two or more digital streams are combined
to facilitate transmission over a common highway. Essentially the process is very simple. If 30
channels, each of 64 kbit/s are to be combined, the width of each pulse is constrained to somewhat less
than 1/30th of the tributary sampling interval which is 1/64 kbit/s. Each of the tributaries is then
delayed by a multiple of 1/30th of this interval so that when the digit streams are combined the
aggregate digit rate has been substantially increased. It might be supposed that this aggregate digit rate
is exactly 30x64 kbit per second, but extra digits must be added to provide ancillary functions, for
example at the receiver channel identification is necessary and signaling information associated with
each channel must be transmitted.
2. 5. Observation of Signals
For all the following experiments the 2/4 channel switch should be set to 2 channels.
The time division multiplexer in this apparatus is a synchronous multiplexer which combines
the digit streams. The basic principles of this operation can be established by observation of the
various signals involved. Using the oscilloscope observe the waveform of the 1 kHz plus 3 kHz
channel 1 generator at the sample and hold input. For ease of observation the sampling pulse source
has been partially synchronized to be channel 1 input and this should be used for triggering, with the
sampling pulse source adjusted to a frequency of about 20 kHz and a pulse width of 10 s . Fine
adjustment of the sampling rate will probably be necessary to lock the pulse stream to the oscilloscope
triggering. To remove any spurious second channel input pulses, that input should be earthed. Without
adjusting the time base of the oscilloscope observation of the time division multiplexer output can be
easily seen to have an envelope identical to that of the original waveform. If necessary, by expansion
of the oscilloscope time base, the repetition period about 50 s and the pulse width of 10 s (at the
base of the pulses) can be clearly observed.

14

Using a signal generator a sinusoid with an amplitude of about 4V and a frequency of 1kHz
can be applied to channel 2 input. Again observing the output, it is clear that two separate pulse
streams exist, one having an envelope consisting of the waveform of channel 1, while the other carries
channel 2. The channel 2 input frequency may need to adjust slightly to ensure a steady trace.
Connecting the multiplexer output to the demultiplexer input and observing each of the channel
outputs allows the demultiplexers fundamental function to be demonstrated. The time delay of the
system can also be measured and the same operations may be performed on channel 2. The frequency
of channel 2 may now be varied and the effective cut-off for a particular sampling frequency can be
measured.
In order to establish the effect of sampling frequency and sample width the following
procedure could be adopted. With the pulse width at 10 s observe the channel 1 output and slowly
reduce the sampling frequency from 20 kHz to its minimum. Using the information established in the
section on the theory of the sampling process, explain in detail the meaning of your observations.
Repeat the process with a sinusoid applied to channel 2 while observing channel 2 output. As only one
frequency component is present in this case it is useful to measure the amplitude of the output signal at
each frequency.
It should be pointed out that as the width of the pulses in the apparatus is increased, the pulse
area also increases. This is because it is convenient in the electronics to use constant height pulses. In
the theory of section 2.3 it is assumed that the area of the pulses is constant and the height adjusts
accordingly. In the theory, it is shown in fig. 2.5 that as width of the pulses increases the height of the
low frequency components in the spectrum stay constant while the higher frequency spectral
components reduce in amplitude. In the experiment, the increase in area of the pulses causes the low
frequency components to increase in amplitude while the high frequency components do not increase
as much. In both cases, of course, the effect of increasing the pulse width is to improve the ratio of
signal to higher order spectra power ratio.
In the experiment the output filters are deliberately made with a cut-off that is not very sharp.
This enables you to clearly see the component in the higher order spectrum interfering with the
original input signal.
Note that a warning light comes on when the pulse width, , exceeds a quarter of the sampling
period. * What is the ratio between pulse width and sample period at which cross talk begins to
become apparent?
* In addition, the commencement of channel overlap(or crosstalk) is indicated by a sudden decrease in
the intensity of the L.E.D. as the pulse width is increased.
2. 6. References

1. P. Bylanski and D.G.W. Ingram, Digital transmission systems, Peter Peregrinus Ltd., 1979.

15

Appendix 1. Convolution
Convolution often comes into statistical analysis when two random variables, such as two
noise waveforms, are to be added together and the statistics of the combination is required. Thinking
in terms of noise is often difficult and a much more colorful example serves to illustrate the principles
just as well and indeed may be more memorable.
The whole school takes part in the annual race, which consists of two legs each of a half mile.
Boys run the first leg and girls run the second leg. No body can run a quarter mile faster than 150
seconds and no boy is slower than 160 seconds. Of the very large number of boys involved in the first
leg there are as many able to run at a chosen speed as there are able to run at any other chosen speed.
The girls are more closely matched in speed, ranging from the fastest 160 seconds to the slowest at
165 seconds. The same equal likelihood at all speeds criterion also applied to the girls. The
partnerships making up each team is done by picking numbers out of a hat, i.e. the combinations of
speeds are random. To find the distribution of the finishing times of the teams, the distribution of
boys times, fig 2.7(a) is convolved with the distribution of the girls times, fig 2.7(b).
The convolution process involves laterally inverting one of the distributions, (see fig. 2.7(b)),
and sliding it past the other distribution. The result of the convolution is a third distribution, whose
value at any particular time T is given by first separating the zero axes of the distributions to be
convolved by T, as shown in fig.2.7(c) and multiplying the distance over which the distributions
overlap by the product of the distribution densities. Thus the resultant density of the convolution of G
and B at a time of 323 seconds is 2 x 1/10 x 1/5. When the distributions do not overlap the resultant is
zero, and for interval over which they completely overlap the resultant is constant at 5 x 1/10 x1/5 as
shown in fig. 2.7(d).
1/10
B

(a)
150

160
1/5
G

(b)
160

165

1/10
B

(c)
150

1/5

160
323

1/10
1/25
(d)
310

Fig. 2.7

16

315

320

325

Gazi University
Department of Electrical and Electronics Engineering
EM437 Communication Systems II Lab.
EXPERIMENT 1 : PCM CODING

Purpose:

To demonstrate the binary coding of d.c. input levels for 3, 4 and 8 bit words.
Experimental Work :

Binary coding of d.c. input levels.

1. Connect the black signal input terminal of the modulator to the blue D.C. Volts terminal,

and set the D.C. Volts to 4 using the oscilloscope.

2. Switch the word length control to 3 Bit, and set the Bit Rate control to approximately mid

range.
3. Connect the oscilloscope Channel 1 input to the red Modulator Output terminal, using

screened leads, with the screen connected to a green terminal (fig 1).

Ch1

Ch2

Fig. 1

4. Switch on the unit and the oscilloscope and allow them to warm up for about 5 minutes.

Then adjust the oscilloscope controls to give the pulses on the display.
5. Vary the D.C. Volts control from 6 to 6, and note that the pulse coding (sequence) on the

display changes. You will find that it is difficult to make sense of the code, the next step
will help.
6. Connect the oscilloscope Channel 2 input to the yellow word Pulse terminal using a

screened lead, with the screen connected to a green terminal (fig 2).

17

Ch1

Ch2

Fig. 2

7. Set the oscilloscope Channel 2 control to 5 V/ div, switch the oscilloscope to trigger from

Ch 2, and adjust the time base to obtain two or three word pulses on the display (fig 3).

Word

Word Pulses

Fig. 3

8. With the display as shown in fig 3, you have now set a Pulse Code Modulation word, as

shown by the Channel 1 trace. With the Word Length control set to 3 Bit, adjust the D.C.
Volts control to alter the bit pattern within the word, giving the sequence of binary codes,
or word levels, shown in fig 4.

Fig. 4

18

Binary Code

Decimal Equivalent

000

100

010

110

001

101

011

111

9. Switch to a 4 bit word. Vary the D.C. Volts control and observe that a new binary

sequence is produced. What are the new codes? Repeat for an 8-bit word.
Conclusion:

The PCM pulse stream contains words of pulses arranged in binary order. The number of
different words depends on the word length.
Equipment List:

Dual beam oscilloscope,

d.c. coupled,
TecQuipment E15f.

19

Gazi University
Department of Electrical and Electronics Engineering
EM437 Communication Systems II Lab.
EXPERIMENT 2 : PCM QUANTISING LEVELS

Purpose:

To demonstrate the quantising levels in 3, 4 and 8 bit pulse code modulation

Experimental Work :

Quantising levels in pulse code modulation.

1. Connect the black signal input terminal of the modulator to the blue D.C. Volts

terminal.
2. Set the D.C. volts to 0 volts. Connect channel 1 of the oscilloscope to the input thus

monitoring the D.C. input signal. Set channel 1 or 2 volts/cm.

3. Switch the word Length control to 3 Bit and set the Bit Rate control to about mid

range.
4. Connect the red output terminal of the Modulator Output to the black terminal of the

Demodulator Input.
5. Connect Channel 2 of the oscilloscope to the red terminal at the Demodulator Output.

Set Channel 2 to 2 volts/cm. The circuit should now look like fig 1.

Ch1

Ch2

Fig. 1

6. Gradually increase the d.c. level eat the input and note that the output from the

demodulator jumps in steps to follow.

20

7. Measure the step size (s) between the output levels, count the number of levels and

note the maximum and minimum values of the voltage ( Vc ).

8. Repeat steps 14 and 15 for a word length (m) of 4 and 8 bits.
Conclusion:

The output from the pulse code modulator can only be at certain d.c. levels. These are the
quantising levels.
For a word of m bits there are 2m quantising levels.
The step size s is given by 2 Vc /( 2m 1) . The step size is least for the 8-bit word and greatest
for the 3-bit word.
Equipment List:

Dual beam oscilloscope,

d.c. coupled,
TecQuipment E15f.

21

Gazi University
Department of Electrical and Electronics Engineering
EM437 Communication Systems II Lab.
EXPERIMENT 3 : PCM QUANTISING NOISE

Purpose:

To observe the quantising noise in pulse code modulation for a sinusoidal input signal.
Experimental Work:
1. Connect the sine wave generator to the black and green Signal Input terminals. Connect

the oscilloscope channel 2 also to these terminals and set the oscilloscope channel 2 also
to these terminals and set the oscilloscope channel 2 to 5 volts/cm.
2. Set the Word Length control to 3-Bit and the Bit Rate to about 40 kHz.
3. Connect the Modulator Output red terminal to the Demodulator Input black terminal.
4. Connect the oscilloscope channel 1 to the Demodulator Output terminals and set channel

1 also to 5 volts/cm. The circuit should now look like fig 1.

Ch1

Ch2

Fig 1

5. Set the oscilloscope time base to 2 ms/div and the sine wave generator to 100 Hz and 10

volts peak-to-peak. Obtain a steady display and superimpose the traces using the position
controls. Trigger from channel 2.

22

6. The output wave from in a stepped form. Check that the step size (s) corresponds to the

measurement in Laboratory Sheet 1.

7. Change the word length to a 4-bit word and note the change in step size. Repeat for an 8-

bit word.
8. Decrease the amplitude of the input signal and note that this does not affect the step size.
9. Increase the amplitude of the input signal to 20 volts peak-to-peak (or beyond if possible)

and note that the output limits at Vc causing clipping of the sine wave.
10. Gradually decrease the Bit Rate to its minimum value of about 4 kHz. Note that the

quantised wave form moves to the right of the input waveform. This is because the
demodulator must receive a complete word before it can decode it. The level displayed
therefore corresponds to the sample taken at the beginning of the previous word. In order
to check this it is necessary to adjust the input frequency very slightly in order to
synchronize the sampling rate with the input frequency.
11. Switch from 3-Bit words to 4-bit words and note that the delay is longer for the 4-bit word

(for the same bit rate). Repeat for an 8-bit word.

12. At first sight it might seem that the step (s) is very much greater at these low bit rates, but

this is not the case, of course. By de-synchronizing the input frequency slightly, note that
the quantised waveform moves through the correct step size but it is the sampling rate that
gives such big steps. If you have the delta modulation experiment (E15e) you should note
that, whereas delta modulation can have samples of any value following each other.
13. With the Bit Rate at 4 kHz and the Word Length at 4 Bit, gradually increase the input

frequency. With a 4-bit word and 4 kHz bit rate the sampling rate is 1 kHz. As
demonstrated in the experiment on sampling and time division multiplex (E15g), the
maximum frequency for correct recovery can only be 500 Hz. Search for synchronization
around 500 Hz and show that the quantised waveform is a square wave. Now increase the
input frequency to around 1 kHz and show that d.c. levels can be obtained for inputs
around 1 kHz. This demonstrates the process of aliasing which is described in detail in
experiment E15g.
14. Set the input frequency to about 50 Hz and set the Bit Rate to its mid range position.

Adjust the oscilloscope to give a convenient trace.

15. Keeping the oscilloscope and the audio generator connected to the Signal Input, connect a

wire from the black Signal Input terminal to the B input of the difference amplifier.
Connect the Demodulator Output red terminal to the A input and the oscilloscope channel
1 to the red output terminal B-A of the difference amplifier as shown in fig2.
23

Ch1

Ch2

Fig 2

16. The oscilloscope trace should now be displaying the quantising noise alone as shown in

fig 3.

Quantising Noise

Fig.3

17. Vary the amplitude of the signal and show that this does not Affect the amplitude of the

quantising noise. Switch the word length and show that this does affect the amplitude of
the quantising noise.

24

Conclusions :

A sinusoidal signal (or any other varying signal) at the input of a pulse code modulator
produces a stepped or quantised waveform at the output.
The sampling rate in pulse code modulation must be at least twice the highest
frequency, otherwise information is lost.
The quantising noise is saw tooth in nature and its amplitude depends on the bits per
word. The quantising noise amplitude is not affected by the signal amplitude or frequency.
Equipment :

Dual beam oscilloscope,

d.c. coupled,
TecQuipment E15f,
Audio frequency sine wave generator with 0 10 V rms output.

25

Gazi University
Department of Electrical and Electronics Engineering
EM437 Communication Systems II Lab.
EXPERIMENT 4: SAMPLING AND TIME DIVISION MULTIPLEX
Purpose:
To investigate the effect of sampling and time division multiplexing upon analog waveforms.
Experimental Work:
Note: For this experiment only channels 1 and 2 are used.
1. Switch on all apparatus and turn the sine wave generator output to zero.
2. Select for two channel operation. Connect channels 2,3 and 4 to the green earth terminal (fig
1).

Ch 1
T DM

TDD

Ch 2
Ch 3
A

B
Ch 4

Fig. 1

3. Set the oscilloscope controls to(say) 1V/cm and 0.2 ms/cm for both inputs. Connect input A of
the oscilloscope to channel 1 input terminal(blue) and synchronise the oscilloscope to channel
1. Use a screened lead and connect the screen to a green earth terminal(fig 1)
4. Observe the internally generated signal(approx. 1kHz plus third harmonic) and check its
fundamental frequency by measurement of the waveform periodic time in the oscilloscope.
5. Using a screened lead with the screen connected to a green terminal, connect input B of the
oscilloscope to the time division multiplexer(TDM) output(fig 1). Set the sample frequency to
20kHz and the sample width to 10 s .
6. Vary the sample width and frequency and observe the relationship between the samples and
the channel 1 waveform. A warning light glows when >T/4 to indicate when the system
limitation is approached. Results are less reliable beyond this point.
Note: In order to relate the channel 1 signal with its samples, the sample frequency and channel 1
signal are synchronised. This causes the channel 1 signal to pull somewhat as the sample
frequency is varied. At certain points this effect changes the shape of the input signal.

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7. Connect the TDM output to the time division demultiplexer(TDD) input. With input A of the
oscilloscope still connected to channel 1 input, connect input B of the oscilloscope to the TDD
channel 1 output.
8. With a sample frequency of 20 kHz and = 10 s , observe the channel 1 output waveform and
compare it with the input waveform. Note any difference between the two waveforms.
9. With kept at 10 s , gradually reduce the sample frequency. Remembering the effect of
synchronisation between channel 1 signal and the samples, note the input and output
waveforms. Note also that the distortion at the output worsens as the sample frequency is
reduced.
10. Remove the earth connection from the black channel 2 input terminal and connect a 4Vp-p 1
kHz sine wave to the channel 2 input(fig 2-see over)
11. Connect input B of the oscilloscope to the TDM output, and set the sample frequency to 20
kHz with = 10 s (fig 2).
12. Observe the TDM output waveform. There is no synchronisation between sample pulses and
channel 2, but fine adjustment of the channel 2 input frequency should produce a sampled sine
wave interleaved between the channel 1 samples.

Ch 1
T DM

TDD

Ch 2

Fig.2
13. With the input A of the oscilloscope on channel 2 input and input B of the oscilloscope on
chanel 2 output, observe waveforms for a sample frequency of 20 kHz and = 10 s .
14. Observe the channel 2 output waveform and gradually reduce the sample frequency from 20
kHz to investigate the aliasing effect. Keep constant 10 s .
15. Note the increasing effect on channel 2 output as the adjacent higher order spectrum comes
within the passband of the channel 2 low pass filter. The effect is illustrated by the idealised
diagram in fig 3.

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16. The interference is a higher frequency sinusoidal wave and its amplitude is compared to the
signal amplitude by the ratio a/s shown in fig 3. To make a measurement of a/s it is best to
trigger the oscilloscope on channel A. It is not easy to measure the frequency of the
interference but for an input frequency of f m and sampling frequency f s it is ( f s - f m ).

Fig. 3
17. With a sampling rate of 10 kHz and = 10 s measure the ratio a/s.
18. Now gradually increase the width until is just less than T/4(indicated by the light and a
jump in the waveform). Measure a/s again noting that the value of a remains constant.
Conclusion:
Analogue signals can be sampled and then recovered provided the sampling frequency is at
least twice the bandwidth of the signal and provided the output filter rejects the adjacent higher order
spectrum.
Sampled signals can be combined into one channel by time division multiplex and separated
by demultiplexing.
The width of the sampling pulses affects the spectrum of the sampled waveform. Wider pulses
reduce the higher frequencies and hence reduce the aliasing noise.
Equipment List:
Dual beam oscilloscope,
d.c. coupled,
TecQuipment E15g.
Audio frequency sine wave generator.

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