1. The experiment demonstrated pulse code modulation (PCM) encoding using an analog-to-digital converter (ADC) and PCM decoding using a digital-to-analog converter (DAC). 2. The sampling frequency determined by the ADC sampling rate generator affected the time between samples on the DAC output. A lower sampling frequency resulted in a longer time between samples and vice versa. 3. The filter cutoff frequency was unaffected by changes to the sampling frequency generator. It was determined solely by the filter capacitor value. With a higher capacitor value, the cutoff frequency decreased.

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)
Cauan, Sarah Krystelle P. September 20, 2011
Signal Spectra and Signal Processing/BSECE 41A1 Score:
Engr. Grace Ramones
Instructor

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

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

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

6. 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
ADC D7
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 measured sampling frequency compare with the frequency of the
sampling frequency generator?
The measured sampling frequency is almost the same 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?
The sampling frequency is more than 20 times of 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 1570796.32 Hz or 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).

8. Question: What is the relationship between the sampling generator output and the DAC
staircase output?
The sampling generator output and the DAC staircase output are both
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 DAC output became analog after filtering. Yes, it is a close
representation of the analog input. The DAC lags the input waveshape.
Step 5 Calculate the cutoff frequency (fC) of the low-pass filter.
Question: How does the filter cutoff frequency compare with the analog input frequency?
They have difference of 5 915.494 Hz.
Step 6 Change the filter capacitor (C) to 20 nF and calculate the new cutoff frequency
(fC).
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

9. Question: How does the time between the samples in Step 8 compare with the time between
the samples in Step 1?
It doubles.
Step 9 Calculate the new sampling frequency (fS) based on the time between the samples
(TS) in Step 8?
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.
Question: How does the new sampling frequency compare with the analog input frequency?
It is 5 times the analog input.

10. 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).
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 more than the time in Step 11.
Step 16 Calculate the new sampling frequency (fS) based on the time between samples
(TS) in Step 15.
Question: How does the new sampling frequency compare with the analog input frequency?
The new sampling frequency is 5 kHz higher than the analog input
frequency.

11. How does the new sampling frequency compare with the Nyquist frequency?
The computed Nyquist frequency is 95199.77 Hz. The Nyquist
frequency is 6.28 times greater than the sampling 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?
The curve plot in Step 17 is the same with the curve plot in Step 13 at
the higher sampling frequency
Step 18 Calculate the frequency of the filter output (f) based on the time period for one
cycle (T).
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 is the same with the frequency of
the analog input. Yes, it is expected.

12. Conclusion:
After conducting the experiment, I conclude that an Analog-to-Digital Converter can be
use for Pulse Code Modulation encoding and Digital-to-Analog Converter is used for Pulse Code
Modulation decoding. The sampling frequency is generated by the ADC. As the frequency of the
sampling frequency generator decreases, the sampling time increases, therefore, the sampling
frequency is inversely proportional to the sampling time of the DAC output. Moreover, the DAC
output waveform is staircase signal. The sampling frequency is always 6.28 times smaller than
the Nyquist frequency. The sampling frequency and the DAC output are both digital.
Meanwhile, the filter cutoff frequency is not affected by the changes in sampling signal
frequency. It is affected by the change in capacitor. As the value of the capacitor increases, the
cutoff frequency decreases. The frequency of the filter is the same with the frequency of the
analog input signal. The waveshape is also the same with the frequency of the analog input
signal.