PRAKTIKUM TELEKOMUNIKASI DIGITAL

LAPORAN SIMULASI
MATLAB SIMULATION OF DELTA
MODULATION TECHNIQUE
Disusun Untuk Memenuhi Tugas
Mata Kuliah Praktikum Telekomunikasi Digital
Semester 4

PEMBIMBING :
Amalia Eka R, ST, MT, M.Sc

Penyusun:
JTD – 2C
Disusun Oleh
Nama : Achmad Farchan Hadi
NIM : 1841160024

PROGRAM STUDI JARINGAN TELEKOMUNIKASI DIGITAL

JURUSAN TEKNIK ELEKTRO
POLITEKNIK NEGERI MALANG
2020
1.1 Objectives
1 To understand the operation theory of Delta modulation
2 To simulate Delta Modulation using Matlab
1.2 Basic Theory
Give a brief explanation about:
1. Analog to Digital Converter
Signals, in general, can be divided into two categories; an analog signal, x(t),
which can be defined in a continuous-time domain and a digital signal, x(n),
which can be represented as a sequence of numbers in a discrete-time domain as
shown in Figure 1.1. The time index n of a discrete-time signal x(n) is an integer
number defined by sampling interval T. Thus, a discrete-time signal, x*(t), can
be represented by a sampled continuous-time signal x(t) as:
∞
n
x∗( t ) = ∑ x ( t ) δ(t−nT )
n=−∞

Where,
δ ( t )=1 , t=0 0,elsewhere
A practical A/D converter transforms x(t) into a discrete-time digital signal,
x*(t), where each sample is expressed with finite precision. Each sample is
approximated by a digital code, i.e., x(t) is transformed into a sequence of finite
precision or quantized samples x(n). This quantization process introduces
Quantization Error in A/D Converters.

Figure 1. 1 Generalized Analog-to-Digital Conversion Process

Most A/D converters can be classified into two groups according to the sampling
rate criteria. Nyquist rate converters, such as a successive approximation register
(SAR), double integration, and oversampling converters, sample analog signals
which have maximum frequencies slightly less than the Nyquist frequency, fN =
fs/ 2, where fs is the sampling frequency. Meanwhile, oversampling converters
perform the sampling process at a much higher rate, fN << Fs (e.g., 64 times for
the DSP56ADC16), where Fs denotes the input sampling rate.

Figure 1. 2 Conventional Analog-to-Digital Conversion Process

illustrates the conventional A/D conversion process that transforms an analog
input signal x(t) into a sequence of digital codes x(n) at a sampling rate of fs =
1/T, where T denotes the sampling interval. Since in Eqn. 2-1 is a periodic
function with period T, it can be represented by a Fourier series given by :
∞

∑ x ( t ) δ(t −nT )
n=−∞

is equivalent to modulating the input signal by carrier signals having

frequencies at 0, fs, 2fs,. . .. In other words, the sampled signal can be expressed
in the frequency domain as the summation of the original signal component and
signals frequency modulated by integer multiples of the sampling frequency as
shown in Figure 2-3. Thus, input signals above the Nyquist frequency, fn,
cannot be properly converted and they also create new signals in the base-band,
which were not present in the original signal. This non-linear phenomenon is a
signal distortion frequently referred to as aliasing .The distortion can only be
prevented by properly lowpass filtering the input signal up to the Nyquist
frequency. This lowpass filter, sometimes called the anti-aliasing filter, must
have flat response over the frequency band of interest (baseband) and attenuate
the frequencies above the Nyquist frequency enough to put them under the noise
floor. Also, the non-linear phase distortion caused by the anti-aliasing filter may
create harmonic distortion and audible degradation. Since the analog anti-
aliasing filter is the limiting factor in controlling the bandwidth and phase
distortion of the input signal, a high performance anti-aliasing filter is required
to obtain high resolution and minimum distortion.
 2. Delta Modulation – Demodulation basic principle
The work on sigma-delta modulation was developed as an extension to
the well established delta modulation. Let’s consider the delta modulation/
demodulation structure for the A/D conversion process. Figure 1.3 shows the
block diagram of the delta modulator and demodulator. Delta modulation is
based on quantizing the change in the signal from sample to sample rather than
the absolute value of the signal at each sample. Since the output of the integrator
in the feedback loop of Figure 1.3(a) tries to predict the input x(t), the integrator
works as a predictor. The prediction error term, , in the current prediction is
quantized and used to make the next prediction. The quantized prediction error
(delta modulation output) is integrated in the receiver just as it is in the feed
back loop. That is, the receiver predicts the input signals as shown in Figure 1.3.
The predicted signal is smoothed with a lowpass filter. Delta modulators,
furthermore, exhibit slope overload for rapidly rising input signals, and their
performance is thus dependent on the frequency of the input signal. In theory,
the spectrum of quantization noise of the prediction error is flat and the noise
level is set by the 1-bit comparator. The Motorola MC3417 Continuously
Variable Slope Delta Modulator operation is based on delta modulation.

Figure 1. 3 Delta Modulation and Demodulation

Here, the integrator circuit contains a delay of Ts. The output of the integrator is
a staircase approximation delayed by Ts. This staircase approximation is compared with
the present sampled input signal at the summer and the difference gives the error signal.
This error signal is given to the quantizer circuit which consists of a hard limiter with an
input-output relationship. Here, the error is quantized into two values i.e. ±δ. Then the
output of the quantizer is coded to produce the desired Delta modulated wave.
At the receiver circuit, demodulation is done using an integrator and a low pass
filter. The modulated wave is first decoded using a decoder and then the staircase
approximation is reconstructed by  passing the positive and negative pulses produced at
the decoder to the integrator. The out-of-band quantizing noise in the high-frequency
staircase waveform is removed by passing the signal through a low-pass filter whose
bandwidth is equal to the original signal bandwidth.
1.3 Experiment Apparatus
1. Matlab software
1.4 Procedures
1. Type the following script onto your M-File
1. clc;
2. clear all
3. close all
4. f=27;%freq in kHz
5. fs=100*f; %sampling frequency
6. t=0:1/fs:2/f;
7. A=1; %amplitude in Volt
8. m=A*sin(2*pi*f*t); %information signal
9. plot(t,m,'k-');
10 hold all;
.
11 d=1/f; %step size
.
12 %% start delta mod
.
13 for n=1:length(m)
.
14 if n==1
.
15 e(n)=m(n);
.
16 eq(n)=d*sign(e(n));
.
17 mq(n)=eq(n);
.
18 else
.
19 e(n)=m(n)-mq(n-1);
.
20 eq(n)=d*sign(e(n));
.
21 mq(n)=mq(n-1)+eq(n);
 .
22 end
.
23 end
.
24 stairs(t,mq,'r-'); %plot staircase approximation of
. information signal m
25
. %% start binary representation of staircase signal
26
.
27 diff=m-mq;
.
28 comp=zeros(1, size(diff,2));
.
29 for i=1:numel(diff)
.

30. if diff(i)>0
31. comp(i)=1;
32. elseif diff(i)<0
33. comp(i)=0;
34. else
35. if i==1
36. comp(i)=0;
37. else
38. comp(i)=comp(i-1);
39. end
40. end
41. end
42. hold on
43. plot(t,comp,'b-') % delta modulated signal
44.
45. grid
on

2. Change the frequency f into your presence number

3. Observe the result. You can differentiate the plot into 3 subplots rather than
plotting in the same figure.
1.5 Measurement Results and discussion
1. What is the characteristics of delta modulation technique?

Figure 1. 4 Result of simulation

Features of Delta Modulation.is :
 An over-sampled input is taken to make full use of the signal correlation.
 The quantization design is simple.
 The input sequence is much higher than the Nyquist rate.
 The quality is moderate.
 The design of the modulator and the demodulator is simple.
 The stair-case approximation of output waveform.
 The step-size is very small, i.e., Δ delta.
 The bit rate can be decided by the user.
 This involves simpler implementation.
Delta Modulation is a simplified form of DPCM technique, also viewed as 1-
bit DPCM scheme. As the sampling interval is reduced, the signal correlation
will be higher. Transfer Characteristics of a delta modulated system follows a
signum function, as it quantizes only two levels and also one-bit at a time. The
two sources of noise in delta modulation are „slope overload‟, when steps are
too small to track the original waveform, and “granularity‟, when steps are too
large.

2. Write the Matlab source code that you used to generate the modulated signal, including
explanation of each code line!
clc;
clear all
close all
f=1;%freq in KHz
fs=100*f; %sampling frequency
t=0:1/fs:2/f; %time
A=1; %amplitude in Volt
m=A*sin(2*pi*f*t); %information signal
subplot(3,1,1) %plotting the signal
plot(t,m,'k-'); %Displays time, signal information waveform, and
color
ylabel('Amplitude (V)') %title of grid y
xlabel('Time(ms)') %title of grid x
title('Information signal') %title of wave
hold all;
d=1/f; %step size

% start delta mod

for n=1:length(m) %looping length of information signal start
from 1
if n==1 %n=1
e(n)=m(n);
eq(n)=d*sign(e(n)); %operation for first step sample
mq(n)=eq(n);
else %other 0
e(n)=m(n)-mq(n-1);
eq(n)=d*sign(e(n)); %operation for n step sample
mq(n)=mq(n-1)+eq(n);
end
end
subplot(3,1,2) %plotting the signal
stairs(t,mq,'r-'); %plot staircas approximation of information
signal m
ylabel('Amplitude (V)') %title of grid y
xlabel('Time(ms)') %title of grid x
title('Approximation of Information Signal m') %title of wave

% start binary representation of staircase signal

diff=m-mq; %taking m length staircase sample for bits display
comp=zeros(1, size(diff,2));
for i=1:numel(diff)
if diff(i)>0 %bit display of 1
comp(i)=1;
elseif diff(i)<0 %bit display of 0
comp(i)=0;
else
if i==1 %bit display of 0
comp(i)=0;
else %bit display of i-1
comp(i)=comp(i-1);
end
end
end
hold on
subplot(3,1,3) %plotting the signal
plot(t,comp,'b-') % delta modulated signal
ylabel('Amplitude (V)') %title of grid y
xlabel('Time(ms)') %title of grid x
title('Delta Modulated Signal') %title of wave
grid on
1.6 Conclusion
It can be concluded from the results of simulations conducted, that Delta
modulation is an analog-to-digital and digital-to-analog signal conversion technique
used to transmit voice information whose quality is not too important. Changes in the
wave to digital, influenced by frequency. So that if analog is lower than the ladder
wave, digital feeding will be perfect.