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Experiment No. 03: Objectives

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0% found this document useful (0 votes)
3 views13 pages

Experiment No. 03: Objectives

Uploaded by

ta1612642434
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
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Experiment No.

03

Pulse Amplitude Modulation (PAM) and Pulse Code Modulation (PCM) and Demodulation
Techniques

OBJECTIVES:

The aim of this experiment is to understand how Pulse Amplitude Modulation (PAM) and Pulse
Code Modulation (PCM) work. We will see how sampling and modulation of a signal are done
and how we can recover the original message. In PAM, we will learn about sampling the
message signal and reconstructing it using a low pass filter. In PCM, we will see how an analog
signal is converted into digital form by sampling, quantization, and encoding. Overall, this
experiment will help us get a better idea about basic communication techniques.

EQUIPMENTS:
 FACET® base unit
 Digital Communications 1 circuit board
 Oscilloscope (dual channel)
 Signal generator (sine wave type)
 Two-post connectors
 Terminal posts

THEORY:

In Pulse Amplitude Modulation (PAM), we take a message signal and create pulses whose
heights (amplitudes) match the signal. These pulses are spaced at regular intervals. If we use a
proper sampling rate (higher than twice the highest frequency of the signal), we can later recover
the original signal using a low pass filter.
In Pulse Code Modulation (PCM), we don’t just create pulses — we convert the signal into
numbers. First, we sample the signal, then we round off (quantize) the values, and finally, we
turn those into binary codes (like 0s and 1s). PCM makes the signal more resistant to noise and
easier to transmit over digital systems like phones and the internet.

PROCEDURE

Part 1: Pulse Amplitude Modulation (PAM)

Sampling the Message Signal

We first connected the 2 kHz message signal (M2) to the input of the sampler. On CH1 of the
oscilloscope, we saw a clean sine wave of 5 V peak-to-peak.

When we connected CH2 to the sample pulse input (SP), we saw a series of narrow sampling
pulses, not PAM yet. This is because these pulses control when sampling happens.

Message signal observation


Sample pulse observation

Sampler output at 8kHz

Measuring Sample Pulse Frequency

By adjusting the sweep to 20 μs/DIV, we measured the time between pulses and calculated the
sampling frequency f_s, which initially came out close to 8 kHz.

Since 8 kHz is greater than twice 2 kHz (the Nyquist limit), this sampling frequency was
sufficient to avoid aliasing.

Observing the PAM Signal

Next, when we connected CH2 to the output of the sampler, we saw the PAM signal — flat-
topped pulses following the shape of the message.

Because the sample pulses captured the message amplitude at each instant, the PAM pulses
matched the instantaneous values of the sine wave. (Insert Figure:

Effect of Different Sampling Frequencies

When we increased the sampling frequency to 16 kHz, the PAM pulses became more frequent,
and the sampled version looked much closer to the original message signal. Higher sampling
gave more detailed information.
Sampled signal at 16 kHz

Then, when we reduced the sampling frequency to 4 kHz, there were fewer pulses per cycle of
the message, and the sampled signal started missing important features.

Sampled signal at 4 kHz

This caused a rougher, less accurate approximation of the message.

Sample and Hold Operation

By switching ON the sample/hold (CM5), the narrow pulses turned into wide, flat steps — a
staircase waveform.
Sampled signal with Sample/Hold at 8kHz sampling rate

Here, the sampled amplitude is held constant until the next sampling instant, which improves the
average energy per sample.

Overlapping Message and PAM Signals

Overlaying the message input (CH1) and the sampler output (CH2) on the oscilloscope showed
that the amplitude of each PAM pulse matched the original sine wave at the moment of
sampling.

Overlapped Input/Output Signal – Message Signal & PAM Output

This proved that the PAM signal accurately carried the information of the message.

(Insert Figure: Overlapped Input/Output Signal - Message Signal and PAM Output)

PAM Demodulation
Low Pass Filter for Reconstruction

After connecting the sampler output to the low pass filter (LPF) and observing the output at 8
kHz sampling, we got a nicely reconstructed sine wave.

The low pass filter removed the high-frequency components and retained the original message
signal.

LPF Output at 8 kHz Sampling Rate

Effect of Lower Sampling Rate

When we reduced the sampling frequency to 4 kHz, even though it was theoretically enough
(Nyquist rate = 4 kHz), practical filter imperfections caused distortion.

LPF Output at 4 kHz Sampling Rate

The filter couldn’t completely remove all unwanted frequencies (like fs-fm components).

Higher Message Frequency with Same Sampling Rate


When we increased the message frequency to 6 kHz but kept sampling at 8 kHz, severe
distortion occurred.

fs = 8 kHz & fm = 6 kHz Reconstruction

This was because the sampling rate was lower than twice the message frequency, causing
aliasing.

Increasing Sampling Rate to 16 kHz

When we sampled at 16 kHz with a message of 3.5 kHz, the reconstructed signal was very good.

A much higher sampling rate avoided aliasing and helped the filter work properly.

fs = 16 kHz & fm = 3.5 kHz Reconstruction

Sample and Hold Effect

With sample-and-hold mode ON at 16 kHz, the recovered signal had slightly higher amplitude.
Demodulation Output with Sample/Hold Sampling

This was because holding the sample gave more average energy to each pulse, boosting the
output amplitude after filtering.

Part B: Pulse Code Modulation (PCM)

Setting up PCM Transmission

We connected CODEC 1 and CODEC 2, and input a 1 kHz sine wave into CODEC 1.

At CODEC 2, we recovered the message and observed that the output closely matched the input.

The PCM system sampled, quantized, and encoded the message digitally.

Ti Input & Output (Reconstructed) Signal at fm = 1 kHz

Observing Frequency Limitations


When we increased the input frequency above 3.5 kHz, the recovered signal was distorted
because the CODEC filter passes only up to 3.5 kHz.

Input & Output (Reconstructed) Signal at fm = 0.1 kHz

Input & Output (Reconstructed) Signal at fm = 4 kHz

Similarly, when we decreased the input below 0.2 kHz, distortion appeared again due to filter
limitations.

Observing PCM Encoding and Timing

We observed the sample pulse SX and the PCM input signal DR at CODEC 2.
Message Signal & Sample Pulse at CODEC-1 Overlap

Sample Pulse at CODEC-1

Timing Pulse at CODEC-2


PCM Input Signal at CODEC-2

Each PCM frame appeared just after a sample pulse, showing the correct timing of the encoding
and transmission.

PCM Decoding and Output Observation

Finally, we saw that the recovered analog output (AR) closely followed the transmitted analog
input (AX) within the designed frequency range.

PCM Input & Analog Output Signal at CODEC-2 Overlap


Analog Input at CODEC-1 and Analog Output at CODEC-2 Overlap

Small differences could be seen due to quantization noise, but overall the signal was
reconstructed well.

DISCUSSION

During the experiment, we faced a few problems that could have caused errors in our results.
One of the main issues was selecting the right sampling rate. If the sampling frequency was too
close to or lower than twice the message signal frequency, the output after demodulation became
distorted due to aliasing. We tried to solve this by always making sure that the sampling rate was
set much higher than the Nyquist rate whenever possible, using the switches on the
communication trainer to change the sampling frequency.

Another problem we noticed was related to oscilloscope triggering. Sometimes the waveforms
were unstable and jittery on the screen. This happened when the trigger source or level was not
set properly. We fixed this by carefully selecting the correct trigger source (either channel 1,
channel 2, or external trigger) depending on what we were observing. We also adjusted the
trigger level until the waveforms became steady and clear.

There were also some distortions seen even when the sampling rate was correctly chosen. This
happened because the low pass filters used in the circuit were not ideal and had gradual cutoffs
instead of perfect sharp cutoffs. As a result, some unwanted frequencies passed through the filter,
affecting the quality of the recovered signal. We tried to minimize this effect by choosing a
sampling rate much higher than the minimum required so that the filter could more easily
separate the message signal from other components.

Finally, during the PCM experiment, small errors sometimes came from slight mismatches
between the transmitted and received signals. This was mainly due to synchronization issues
between the timing pulses of CODEC 1 and CODEC 2. However, by carefully observing the
sample and timing pulses on the oscilloscope and adjusting the signal generator properly, we
were able to maintain synchronization and get a clean recovered signal most of the time.

Overall, by making small corrections during the experiment and understanding the reason behind
each error, we were able to get reasonably accurate and expected results.

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