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)
Pagara, Sheila Marie P. 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 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.

5. 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.
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

6. 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
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

7. Question: How did the measure sampling frequency compare with the frequency of the
sampling frequency generator?
Both have difference of 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 higher. It is more than twice the analog input frequency.
How did the sampling frequency compare with the Nyquist frequency?
The Nyquist frequency is 6.28 times 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?
They are 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 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 after filtering. Yes it is close representation.
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?

8. The time 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.
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.
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

9. 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?
The frequency of the filter output and the frequency of the analog input are the
same. For sampling frequency of 15.037 kHz, it is expected to have same frequency.

10. CONCLUSION:
I conclude that 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. 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.