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)
Agdon, Berverlyn B. 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

5. 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.
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

6. 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
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.

7. 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?
They have a 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 the analog input frequency. Yes it is more than twice the analog input
frequency.
How did the sampling frequency compare with the Nyquist frequency?
The 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?
The staircase output and the sampling generator output are both in digital
form
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

8. Questions: What happened to the DAC output after filtering? Is the filter output
waveshape a close representation of the analog input waveshape?
The DAC output became analog after it was being filtered. 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.
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.

9. 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
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.

10. 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 filter output frequency is the same with the analog input. Yes, it is
expected based on the sampling frequency

11. CONCLUSION:
After conducting the experiment, I conclude PCM coding can be demonstrated
through ADC while DAC is used for PCM decoding. Also, based on the output graph displayed
by the oscilloscope, the staircase output frequency is almost the same with the sampling
frequency generator, and is several times higher than the input analog frequency. The
sampling time is inversely proportional to the sampling frequency. The output of the filter is
like the input analog signal. Its frequency is also the same with the input analog frequency.
The cutoff frequency remains constant even if the sampling generator frequency changes but
is inversely proportional to the capacitance and resistance. Finally, the Nyquist frequency is
always 6.28 times higher than the sampling frequency generated by the ADC.