1. NATIONAL COLLEGE OF SCIENCE & TECHNOLOGY
Amafel Bldg. Aguinaldo Highway Dasmariñas City, Cavite
EXPERIMENT 2
DIGITAL COMMUNICATION OF ANALOG DATA USING
PULSE-CODE MODULATION (PCM)
Pula, Rolando A. September 20, 2011
Signal Spectra and Signal Processing/BSECE 41A1 Score:
Engr. Grace Ramones
Instructor

2. Objectives:
Demonstrate PCM encoding using an analog-to-digital converter (ADC).
Demonstrate PCM encoding using an digital-to-analog converter (DAC)
Demonstrate how the ADC sampling rate is related to the analog signal
frequency.
Demonstrate the effect of low-pass filtering on the decoder (DAC) output.

3. Sample Computation
Step2
Step 6
Step 9
Step 12
Step 14
Step 16
Step 18

4. Data Sheet:
Materials
One ac signal generator
One pulse generator
One dual-trace oscilloscope
One dc power supply
One ADC0801 A/D converter (ADC)
One DAC0808 (1401) D/A converter (DAC)
Two SPDT switches
One 100 nF capacitor
Resistors: 100 Ω, 10 kΩ
Theory
Electronic communications is the transmission and reception of information
over a communications channel using electronic circuits. Information is defined
as knowledge or intelligence such as audio voice or music, video, or digital
data. Often the information id unsuitable for transmission in its original form and
must be converted to a form that is suitable for the communications system.
When the communications system is digital, analog signals must be converted
into digital form prior to transmission.
The most widely used technique for digitizing is the analog information signals
for transmission on a digital communications system is pulse-code modulation
(PCM), which we will be studied in this experiment. Pulse-code modulation
(PCM) consists of the conversion of a series of sampled analog voltage levels
into a sequence of binary codes, with each binary number that is proportional
to the magnitude of the voltage level sampled. Translating analog voltages
into binary codes is called A/D conversion, digitizing, or encoding. The device
used to perform this conversion process called an A/D converter, or ADC.
An ADC requires a conversion time, in which is the time required to convert
each analog voltage into its binary code. During the ADC conversion time, the
analog input voltage must remain constant. The conversion time for most
modern A/D converters is short enough so that the analog input voltage will not
change during the conversion time. For high-frequency information signals, the
analog voltage will change during the conversion time, introducing an error
called an aperture error. In this case a sample and hold amplifier (S/H amplifier)
will be required at the input of the ADC. The S/H amplifier accepts the input

5. and passes it through to the ADC input unchanged during the sample mode.
During the hold mode, the sampled analog voltage is stored at the instant of
sampling, making the output of the S/H amplifier a fixed dc voltage level.
Therefore, the ADC input will be a fixed dc voltage during the ADC conversion
time.
The rate at which the analog input voltage is sampled is called the sampling
rate. The ADC conversion time puts a limit on the sampling rate because the
next sample cannot be read until the previous conversion time is complete. The
sampling rate is important because it determines the highest analog signal
frequency that can be sampled. In order to retain the high-frequency
information in the analog signal acting sampled, a sufficient number of samples
must be taken so that all of the voltage changes in the waveform are
adequately represented. Because a modern ADC has a very short conversion
time, a high sampling rate is possible resulting in better reproduction of
high0frequency analog signals. Nyquist frequency is equal to twice the highest
analog signal frequency component. Although theoretically analog signal can
be sampled at the Nyquist frequency, in practice the sampling rate is usually
higher, depending on the application and other factors such as channel
bandwidth and cost limitations.
In a PCM system, the binary codes generated by the ADC are converted into
serial pulses and transmitted over the communications medium, or channel, to
the PCM receiver one bit at a time. At the receiver, the serial pulses are
converted back to the original sequence of parallel binary codes. This
sequence of binary codes is reconverted into a series of analog voltage levels
in a D/A converter (DAC), often called a decoder. In a properly designed
system, these analog voltage levels should be close to the analog voltage
levels sampled at the transmitter. Because the sequence of binary codes
applied to the DAC input represent a series of dc voltage levels, the output of
the DAC has a staircase (step) characteristic. Therefore, the resulting DAC
output voltage waveshape is only an approximation to the original analog
voltage waveshape at the transmitter. These steps can be smoothed out into
an analog voltage variation by passing the DAC output through a low-pass
filter with a cutoff frequency that is higher than the highest-frequency
component in the analog information signal. The low-pass filter changes the
steps into a smooth curve by eliminating many of the harmonic frequency. If
the sampling rate at the transmitter is high enough, the low-pass filter output
should be a good representation of the original analog signal.

6. In this experiment, pulse code modulation (encoding) and demodulation
(decoding) will be demonstrated using an 8-bit ADC feeding an 8-bit DAC, as
shown in Figure 2-1. This ADC will convert each of the sampled analog voltages
into 8-bit binary code as that represent binary numbers proportional to the
magnitude of the sampled analog voltages. The sampling frequency
generator, connected to the start-of conversion (SOC) terminal on the ADC, will
start conversion at the beginning of each sampling pulse. Therefore, the
frequency of the sampling frequency generator will determine the sampling
frequency (sampling rate) of the ADC. The 5 volts connected to the VREF+
terminal of the ADC sets the voltage range to 0-5 V. The 5 volts connected to
the output (OE) terminal on the ADC will keep the digital output connected to
the digital bus. The DAC will convert these digital codes back to the sampled
analog voltage levels. This will result in a staircase output, which will follow the
original analog voltage variations. The staircase output of the DAC feeds of a
low-pass filter, which will produce a smooth output curve that should be a close
approximation to the original analog input curve. The 5 volts connected to the
+ terminal of the DAC sets the voltage range 0-5 V. The values of resistor R and
capacitor C determine the cutoff frequency (fC) of the low-pass filter, which is
determined from the equation
Figure 23–1 Pulse-Code Modulation (PCM)
XSC2
G
T
A B C D
S1 VCC
Key = A 5V
U1
Vin D0
S2
D1
V2 D2
D3 Key = B
2 Vpk D4
10kHz
D5
0° Vref+
D6
Vref-
D7
SOC VCC
OE EOC 5V
D0
D1
D2
D3
D4
D5
D6
D7
ADC
V1 Vref+ R1
VDAC8 Output
5V -0V Vref- 100Ω
200kHz
U2
R2
10kΩ C1
100nF

7. In an actual PCM system, the ADC output would be transmitted to serial format
over a transmission line to the receiver and converted back to parallel format
before being applied to the DAC input. In Figure 23-1, the ADC output is
connected to the DAC input by the digital bus for demonstration purposes only.
PROCEDURE:
Step 1 Open circuit file FIG 23-1. Bring down the oscilloscope
enlargement. Make sure that the following settings are selected.
Time base (Scale = 20 µs/Div, Xpos = 0 Y/T), Ch A(Scale 2 V/Div,
Ypos = 0, DC) Ch B (Scale = 2 V/Div, Ypos = 0, DC), Trigger (Pos
edge, Level = 0, Auto). Run the simulation to completion. (Wait
for the simulation to begin). You have plotted the analog input
signal (red) and the DAC output (blue) on the oscilloscope.
Measure the time between samples (TS) on the DAC output
curve plot.
TS = 4 µs
Step 2 Calculate the sampling frequency (fS) based on the time
between samples (TS)
fS = 250 kHz
Question: How did the measure sampling frequency compare with the
frequency of the sampling frequency generator?
The difference is 50 kHz.
How did the sampling frequency compare with the analog input frequency?
Was it more than twice the analog input frequency?
It is 20 times the analog input frequency. It is more than twice
the analog input frequency.
How did the sampling frequency compare with the Nyquist frequency?
The Nyquist frequency is higher. Nyquist is 6.28 times more than
the sampling frequency.
Step 3 Click the arrow in the circuit window and press the A key to change
Switch A to the sampling generator output. Change the oscilloscope
time base to 10 µs/Div. Run the simulation for one oscilloscope screen
display, and then pause the simulation. You are plotting the sampling
generator (red) and the DAC output (blue).
Question: What is the relationship between the sampling generator output and
the DAC staircase output?
Both outputs are both in digital
Step 4 Change the oscilloscope time base scale to 20 µs/Div. Click the arrow
in the circuit window and press the A key to change Switch A to the

8. analog input. Press the B key to change the Switch B to Filter Output.
Bring down the oscilloscope enlargement and run the simulation to
completion. You are plotting the analog input (red) and the low-pass
filter output (blue) on the oscilloscope
Questions: What happened to the DAC output after filtering? Is the filter output
waveshape a close representation of the analog input waveshape?
The output became analog. Yes.
Step 5 Calculate the cutoff frequency (fC) of the low-pass filter.
fC = 15.915 kHz
Question: How does the filter cutoff frequency compare with the analog input
frequency?
They have difference of approximately 6 kHz.
Step 6 Change the filter capacitor (C) to 20 nF and calculate the new cutoff
frequency (fC).
fC = 79.577 kHz
Step 7 Bring down the oscilloscope enlargement and run the simulation to
completion again.
Question: How did the new filter output compare with the previous filter output?
Explain.
It is almost the same.
Step 8 Change the filter capacitor (C) back to 100 nF. Change the Switch B
back to the DAC output. Change the frequency of the sampling
frequency generator to 100 kHz. Bring down the oscilloscope
enlargement and run the simulation to completion. You are plotting
the analog input (red) and the DAC output (blue) on the oscilloscope
screen. Measure the time between the samples (TS) on the DAC
output curve plot (blue)
TS = 9.5µs
Question: How does the time between the samples in Step 8 compare with the
time between the samples in Step 1?
The time between the samples in Step 8 doubles.
Step 9 Calculate the new sampling frequency (fS) based on the time
between the samples (TS) in Step 8?
fS=105.26Hz
Question: How does the new sampling frequency compare with the analog
input frequency?
It is 10 times the analog input frequency.

9. Step 10 Click the arrow in the circuit window and change the Switch B to the
filter output. Bring down the oscilloscope enlargement and run the
simulation again.
Question: How does the curve plot in Step 10 compare with the curve plot in
Step 4 at the higher sampling frequency? Is the curve as smooth as in
Step 4? Explain why.
Yes, they are the same. It is as smooth as in Step 4. Nothing changed.
It does not affect the filter.
Step 11 Change the frequency of the sampling frequency generator to 50 kHz
and change Switch B back to the DAC output. Bring down the
oscilloscope enlargement and run the simulation to completion.
Measure the time between samples (TS) on the DAC output curve plot
(blue).
TS = 19µs
Question: How does the time between samples in Step 11 compare with the
time between the samples in Step 8?
It doubles.
Step 12 Calculate the new sampling frequency (fS) based on the time
between samples (TS) in Step 11.
fS=52.631 kHz
Question: How does the new sampling frequency compare with the analog
input frequency?
It is 5 times the analog input.
Step 13 Click the arrow in the circuit window and change the Switch B to the
filter output. Bring down the oscilloscope enlargement and run the
simulation to completion again.
Question: How does the curve plot in Step 13 compare with the curve plot in
Step 10 at the higher sampling frequency? Is the curve as smooth as in
Step 10? Explain why.
Yes, nothing changed. The frequency of the sampling generator does
not affect the filter.
Step 14 Calculate the frequency of the filter output (f) based on the period for
one cycle (T).
T=10kHz
Question: How does the frequency of the filter output compare with the
frequency of the analog input? Was this expected based on the
sampling frequency? Explain why.
It is the same. Yes, it is expected.

10. Step 15 Change the frequency of the sampling frequency generator to 15 kHz
and change Switch B back to the DAC output. Bring down the
oscilloscope enlargement and run the simulation to completion.
Measure the time between samples (TS) on the DAC output curve plot
(blue)
TS = 66.5µs
Question: How does the time between samples in Step 15 compare with the
time between samples in Step 11?
It is 3.5 times higher than the time in Step 11.
Step 16 Calculate the new sampling frequency (fS) based on the time
between samples (TS) in Step 15.
fS=15.037 kHz
Question: How does the new sampling frequency compare with the analog
input frequency?
It is 5 kHz greater than the analog input frequency.
How does the new sampling frequency compare with the Nyquist frequency?
It is 6.28 times smaller than the Nyquist frequency.
Step 17 Click the arrow in the circuit window and change the Switch B to the
filter output. Bring down the oscilloscope enlargement and run the
simulation to completion again.
Question: How does the curve plot in Step 17 compare with the curve plot in
Step 13 at the higher sampling frequency?
They are the same.
Step 18 Calculate the frequency of the filter output (f) based on the time
period for one cycle (T).
f=10kHz
Question: How does the frequency of the filter output compare with the
frequency of the analog input? Was this expected based on the
sampling frequency?
They are the same. Yes it is expected for that sampling frequency.

11. CONCLUSION:
I can able to say that the Analog to Digital and Digital to Analog Converters
can be use as Pulse Code Modulation encoder and decoder.
Based on the circuit performed, DAC output is a staircase while the filter
output is analog. The staircase output sampling time is inversely proportional to the
sampling frequency while the filter frequency is always equal with the analog input
frequency. The cutoff frequency of the filter is inversely proportional to the
capacitance. Lastly, the Nyquist is always 6.28 times higher than the DAC output
frequency.