Pcs Exp 5
Pcs Exp 5
: 05
TITLE: PCM
EQUIPMENT REQUIRED: Uniform PCM Modem kit, Spectrum Analyzer, CRO, and
Connecting Wires.
THEORY:
PCM
A signal can be represented in Analog or Digital from. An analog signal is characterized by the
fact that its attributes (like: amplitude, frequency and phase) can take any value over a
continuous range. On the other hand, digital signals are discrete in nature and can take only
discrete and finite values. One can convert an analog signal to a digital signal by sampling and
quantizing (collectively called analog-to-digital conversion, or ADC). It is typically more
efficient to process the resulting discrete signals by digital signal processors. The processed
signals are then converted back into analog signals using a reconstruction or interpolation
operation (called digital-to-analog conversion, or DAC).
In analog communication of continuous wave (CW) modulation one of the parameter of
sinusoidal carrier wave is varied according to modulation signal amplitude. But in Pulse code
modulation (PCM) is a discrete time and discrete amplitude representation for a signal and it has
no CW counterpart. Hence PCM is known as digital to analog conversion process or its code the
continuous wave into digital from and hence known as code conversion technique
Process of converting analog signal into digital signal and transmission or PCM is shown in
figure. The steps are sampling, quantization, encoding and parallel to serial conversion.
Incoming signal is passed through a LPF that acts as an anti-aliasing filter, which limits the
frequency of incoming signal to fs/2. Sampling process uses sampling clock which is a train of
pulses with frequency fs and smaller duty cycle. This is shown in figure
Fig.5.1 : PCM Transmitter
PCM receiver is shown in figure below. The correctly received bits are applied to the parallel to
serial conversion which accepts „n‟ bit digital data serially and converts it into parallel form.
These bits are applied to the decoder (D-A convertor) which generates a pulse corresponding to
the received „n‟ bits. Thus output of the decoder is the quantized PAM pulses. These pulses
contain quantization error which is very small compared to pulse amplitude and it cannot be
removed. This PAM signal is passed through LPF acting as interpolation filter for reconstructing
original signal from sampled waveform.
Fig.5.3 : PCM Receiver
Quantization Process:
The 2‟nd step in the PCM system is to quantize the modulating signal. The modulating signal can
assume an infinite no. of different level between the two limit values, which define the range of
the signal. In PCM a coded no is transmitted for each level sampled in the modulating signal. If
the exact no corresponding to the exact voltage were to be transmitted for every sample, an
infinitely large no of different code symbols would be needed. Quantization has the effect of
reducing this infinite no of levels to a relatively small number, which can be coded without
difficulty.
In the quantization process, the total range of the modulating signal is divided into a no of small
sub ranges. The number will depend on the nature of the modulating signal and will form as few
as 8 to as many as 1024 levels. A number that is an integer power of two is usually chosen
because of the ease of generating binary codes. The result is stepped waveform, which follows
the counter of the original modulating signal with each step synchronized to the sampling period.
The quantized staircase waveform is an approximation to the original waveform. The difference
between the two-wave form amounts to “noise” added to the signal by the quantizing circuit. The
mean square quantization noise voltage has a value of E2np = a2/12 Where „a‟ is the voltage of
each step. As a result the number of quantization levels must be kept high in order to keep the
quantization noise below some acceptable limit given by the power signal-to-noise ratio, which
is the ratio of average noise power. In this process of quantization the difference between two
quantization levels i.e. step size remains constant throughout the amplitude range and hence it is
known as linear or uniform quantization and PCM is known as linear PCM.
Fig.5.4: Example of quantization process considering 8 levels
PROCEDURE:
PCM generation:
1) Connect MOD I/P terminal of PCM modulator to SINE O/P terminal of audio oscillator.
Connect CRO channel -1 at SINE O/P terminal of audio oscillator. Connect ground of
probe to ground terminal of audio oscillator. Adjust amplitude of SINE wave to 2 Vp-p
and audio frequency to 1 KHz. Waveform T1.
2) Connect CRO channel -2 at Sampling Clock terminal (T2) of sampling pulse generator.
Connect ground of probe to ground terminal of PCM demodulator. Keep frequency
selection switch at FAST position. Waveform T2.
3) Connect CRO channel -2 at PCM O/P terminal of modulator. Keep bit select switch at 4-
bit position.
4) The PCM modulated waveform will be seen. Waveform T3.
5) Observe Ramp signal Waveform T6.
PCM demodulation:
1) Connect link between PCM O/P terminal & PCM I/P terminal. Connect CRO channel -2
at RAW DATA terminal of demodulator. Observe quantized recovered raw data signal
Waveform T4.
2) Change bit selection switch to 3-bit position & observe steps in output.the steps will be
reduced to 8 from 16.
3) Then connect CRO channel -2 at FILTER O/P terminal of LPF and observe filtered
recovered output signal WaveformT5.
Effect of DC signal input on PCM output:
1) Connect variable DC signal at MOD I/P terminal of PCM modulator. Connect ground of
probe to ground terminal of audio oscillator. Keep frequency selection switch at FAST
position Waveform T7.
2) See the counting effect on LEDs o/p.
OBSERVATIONS:
1) Observation table:
CALCULATIONS:
1. For 3-bit PCM
Bit rate = n*fs = 3*(25KHz) =75Kbps
Bandwidth = n*fs/2= 75/2 = 37.5 KHz
SNR (db) = 1.8 +6(n) = 1.8+ 6(3) =19.8 db
2. Similarly for 4-bit PCM
Bit rate = n*fs = 4*(25KHz) =100 Kbps
Bandwidth = n*fs/2 = 100/2 = 50 KHz
SNR (db) = 1.8 +6(n) = 1.8+ 6(4) =25.8 db
PROGRAM:
clc;
close;
clear;
f=2;
fs =20* f;// Sampling Frequency
t= 0:1/ fs:2;
a=2;
msg= a* sin (2.* %pi* f* t);
subplot (3 ,1 ,1);
plot(t, msg)
xlabel( "TIME " );
ylabel( "AMPLITUDE " )
title( "Message S i g n a l " );
PROGRAM:
clear all;
clc;
n = input('Enter the number of bits to encode: ')
W = input('Enter the message signal bandwidth: ')
B = n*W;
disp('Channel Width in Hertz: ',B)
SNRo = 6*n - 7.2;
//SNRo = 4.8 - 6*n;
//SNRo = 1.8 + 6*n;
disp(' Output Signal to Noise Ratio in dB : ',SNRo)