10 Exp

Differential Pulse Code
Modulation Trainer
ST2113
Learning Material
Ver 1.2
An ISO 9001:2008 company

Scientech Technologies Pvt. Ltd.
94, Electronic Complex, Pardesipura, Indore - 452 010 India,
+ 91-731 4211100, : info@scientech.bz , : www.ScientechWorld.com
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Scientech Technologies Pvt. Ltd. 2

ST2113
Differential Pulse Code Modulation Trainer

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Table of Contents
1. Safety Instructions 4
2. Introduction 5
3. Features 7
4. Technical Specifications 8
5. Theory 9
6. Differential Pulse Code Modulation 33
7. ST2113 DPCM Trainer- Brief Description 38
8. Experiments
• Experiment 1 43
Study of Differential Pulse Code Modulation and Demodulation
Technique
• Experiment 2 46
To verify experimentally that DPCM is a Differentiation Process
• Experiment 3 49
To establish voice link using DPCM Technique
9. Frequently Asked Questions 51
10. Warranty 57
11. List of Contents 57

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Safety Instructions
Read the following safety instructions carefully before operating the instrument. To
avoid any personal injury or damage to the instrument or any product connected to it.
Do not operate the instrument if suspect any damage to it.
The instrument should be serviced by qualified personnel only.
For your safety:

Use proper Mains cord : Use only the mains cord designed for this instrument.
Ensure that the mains cord is suitable for your
country.
Ground the Instrument : This instrument is grounded through the protective
earth conductor of the mains cord. To avoid electric
shock the grounding conductor must be connected to
the earth ground. Before making connections to the
input terminals, ensure that the instrument is properly
grounded.
Observe Terminal Ratings : To avoid fire or shock hazards, observe all ratings and
marks on the instrument.
Use only the proper Fuse : Use the fuse type and rating specified for this
instrument.
Use in proper Atmosphere : Please refer to operating conditions given in the
manual.
1. Do not operate in wet / damp conditions.
2. Do not operate in an explosive atmosphere.
3. Keep the product dust free, clean and dry.

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Introduction
ST2113, Differential Pulse Code Modulation Trainer is a manifestation of our
increasing efforts to present the modern technology in a best way to the people who
want to unfold the mysteries behind ever increasing communication super highway.
To present it in a best way the trainer incorporates the practical operating frequencies
for sampling, audio processing and data processing that are commonly used in our
public telephone networks.
ST2113, DPCM trainer comprises of following major blocks :
• Audio Signal generator (with sine and square wave output).
• Modulator block which consists of :
a. Difference amplifier with sign manipulation circuit
b. ADC
c. Parallel to serial converter
d. Sample Predictor
• Demodulator block which consists of :
a. Serial to parallel converter
b. Adder and subtractor units
c. Bus latches
d. DAC
e. Output Low pass filter
• Audio input processing circuit.
• Audio output processing circuit.
• Clock and entire control Signal section.

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Features
• Onboard DPCM Transmitter and receiver
• Onboard Signal generator block
• Onboard Audio input & output processing circuits
• Clock and entire control Signal section
• Detailed signal processing circuit with complete data and control signal
flow

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Technical Specifications
Signal generator block
Functions : Sine and Square
O/P frequency range : 300Hz to 3.4 KHz
Audio blocks : Audio I/P and O/P processing circuits
Control signals : R/W for ADC, reset. Latch enables, OEs.
Sampling frequency : 8 KHz
Total Bits per sample : 5 bits including sign bit
Bandwidth improvement
Compared to 8 bit PCM : 3 bits per sample
Interconnections : 2mm socket
Power Supply : ± 5V, ± 12V DC, 200mA
Dimensions (mm) : W325, H90, D255
Weight : 1 Kg. (approximately)

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Theory
Pulse Code Modulation (PCM) is an extension of PAM wherein each analogue
sample value is quantized into a discrete value for representation as a digital code
word.
Thus, as shown below, a PAM system can be converted into a PCM system by adding
a suitable analogue-to-digital (A/D) converter at the source and a digital-to-analogue
(D/A) converter at the destination.
PCM is a true digital process as compared to PAM. In PCM the speech signal is
converted from analogue to digital form.
PCM is standardized for telephony by the ITU-T (International Telecommunications
Union - Telecoms, a branch of the UN), in a series of recommendations called the G
series. For example the ITU-T recommendations for out-of-band signal rejection in
PCM voice coders require that 14 dB of attenuation is provided at 4 kHz. Also, the
ITU-T transmission quality specification for telephony terminals requires that the
frequency response of the handset microphone has a sharp roll-off from 3.4 kHz.
In quantization the levels are assigned a binary codeword. All sample values falling
between two quantization levels are considered to be located at the centre of the
quantization interval. In this manner the quantization process introduces a certain
amount of error or distortion into the signal samples. This error known as quantization
noise is minimized by establishing a large number of small quantization intervals. Of
course, as the number of quantization intervals increase, so must the number or bits
increase to uniquely identify the quantization intervals. For example, if an analogue
voltage level is to be converted to a digital system with 8 discrete levels or
quantization steps three bits are required. In the ITU-T version there are 256
quantization steps, 128 positive and 128 negative, requiring 8 bits. A positive level is
represented by having bit 8 (MSB) at 0, and for a negative level the MSB is 1.
Pulse code modulation, more popularly known as PCM is the most widely used digital
modulation system. It is a widely known fact that the analog modulation systems are
most prone to the noise present in the channel and receiver. As we will see further that
the digital modulation systems are far less sensitive to noise as compared to analog
modulation. The basis of digital modulation systems lies on pulse modulation i.e. a
particular characteristic of the pulse is varied in accordance with the information
signal.
In Pulse Modulation, analog message is transmitted in discrete time. First of all,
sampling of the message signal should be performed. Considering the sampling
process, the sampled signal appears as a train of samples which is a form of PAM
(Pulse Amplitude Modulation) signal. When M levels are used to quantize this signal,
this modulation is called M-PAM. If those pulses were converted to digital numbers,
then the train of numbers so generated would be called as Pulse Code Modulated –
PCM signal. In PCM, modulation process is executed in three steps:
1. Sampling
2. Quantizing
3. Coding
These steps are shown in Figure 1 with a block diagram:

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Figure 1
PCM block Diagram

Figure 2
In PCM, the information signal x(t) is first sampled with the appropriate
sampling frequency (sampling frequency fs ≥ 2×highest frequency of the
information signal (fx) ), then the sampled levels are quantized to appropriate
quantization levels. In the last step, each quanta level is demonstrated by a
two-code word, that is by a finite number of {0,1} sequence. After this step,
the signal is called as PCM wave.

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Pulse Code Modulation

Steps in Pulse Code Modulation:
Sampling:
The signals which are required to be transmitted as information is known as
information signal and in the case of voice communication this will be a continuously
changing signal containing speech information. The aim of the kit is to transmit the
signals in digital form and is to reproduce this information signal in analog form at the
receiving end of the communication system with the help of sampling and
reconstruction trainer.
In the exercises to follow, you will simulate audio signal by a 1 kHz test signal
provided On-board. The repetitive, non-changing waveform does not contain
information. Provided the frequency of the test-signal lies within the frequency range
which an information signal will occupy, a test signal of this type can be extremely
helpful in system analysis and testing.
The voice signals are limited to the range 300 Hz to 3.4 kHz, a 1 kHz frequency fits
conveniently in this range and can be used to demonstrate and test many techniques
used in communication system.
Theory of sampling:
The signals we use in the real world, such as our voice, are called "analog" signals.
To process these signals for digital communication, we need to convert analog signals
to "digital" form. While an analog signal is continuous in both time and amplitude, a
digital signal is discrete in both time and amplitude. To convert continuous time
signal to discrete time signal, a process is used called as sampling. The value of the
signal is measured at certain intervals in time. Each measurement is referred to as a
sample.
Principle of sampling:
Consider an analogue signal x(t) that can be viewed as a continuous function of time,
as shown in figure 3. We can represent this signal as a discrete time signal by using
values of x(t) at intervals of nTs to form x(nTs) as shown in figure 3. We are
"grabbing" points from the function x(t) at regular intervals of time, Ts, called the
sampling period.

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Basic Sampling Process

Figure 3
Sampling of signal at sampling interval (period) Ts

Figure 4
Figure 4 depicts the sampling of a signal at regular interval (period) t=nTs where n is
an integer. The sampling signal is a regular sequence of narrow pulses δ (t) of
amplitude 1.Figure 5 shows the sampled output of narrow pulses δ (t) at regular
interval of time.

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Sampled Output of narrow pulses δ (t)

Figure 5
The time distance Ts is called sampling interval or sampling period, fs=1/Ts is called
as sampling frequency (Hz or samples/sec), also called sampling rate.
The Sampling Theorem:
The Sampling Theorem states that a signal can be exactly reproduced if it is sampled
at a frequency Fs, where Fs is greater than twice the maximum frequency Fmax in the
signal.
Fs > 2· Fmax
The frequency 2· Fmax is called the Nyquist sampling rate. Half of this value, Fmax, is
sometimes called the Nyquist frequency.
The sampling theorem is considered to have been articulated by Nyquist in 1928 and
mathematically proven by Shannon in 1949. Some books use the term "Nyquist
Sampling Theorem", and others use "Shannon Sampling Theorem". They are in fact
the same sampling theorem.
The sampling theorem clearly states what the sampling rate should be for a given
range of frequencies. In practice, however, the range of frequencies needed to
faithfully record an analog signal is not always known beforehand. Nevertheless,
engineers often can define the frequency range of interest. As a result, analog filters
are sometimes used to remove frequency components outside the frequency range of
interest before the signal is sampled.
For example, the human ear can detect sound across the frequency range of 20 Hz to
20 kHz. According to the sampling theorem, one should sample sound signals at least
at 40 kHz in order for the reconstructed sound signal to be acceptable to the human
ear. Components higher than 20 kHz cannot be detected, but they can still pollute the
sampled signal through aliasing. Therefore, frequency components above 20 kHz are
removed from the sound signal before sampling by a band-pass or low-pass analog
filter.

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Nyquist Criterion
As shown-in the figure 6 the lowest sampling frequency that can be used without the
sidebands overlapping is twice the highest frequency component present in the
information signal. If we reduce this sampling frequency even further, the sidebands
and the information signal will overlap and we cannot recover the information signal
simply by low pass filtering. This phenomenon is known as fold-over distortion or
aliasing.
Nyquist Criterion (Sampling Theorem)

Figure 6
The Nyquist criteria states that a continuous signal band limited to Fm Hz can be
completely represented by and reconstructed from the samples taken at a rate greater
than or equal to 2Fm samples/second.
This minimum sampling frequency is called as Nyquist Rate i.e. for faithful
reproduction of information signal fs > 2 fm.
For audio signals the highest frequency component is 3.4 KHz.

So, Sampling Frequency ≥ 2 fm
≥ 2 x 3.4 KHz
≥ 6.8 KHz
Practically, the sampling frequency is kept slightly more than the required rate. In
telephony the standard sampling rate is 8 KHz. Sample quantifies the instantaneous
value of the analog signal point at sampling point to obtain pulse amplitude output.

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Nyquist’s Uniform Sampling Theorem for Low pass Signal:

Part - I If a signal x(t) does not contain any frequency component beyond W Hz, then
the signal is completely described by its instantaneous uniform samples with sampling
interval (or period ) of Ts < 1/(2W) sec.
Part – II The signal x(t) can be accurately reconstructed (recovered) from the set of
uniform instantaneous samples by passing the samples sequentially through an ideal
(brick-wall) low pass filter with bandwidth B, where W ≤ B < fs – W and fs = 1/(Ts).
As the samples are generated at equal (same) interval (Ts) of time, the process of
sampling is called uniform sampling. Uniform sampling, as compared to any non-
uniform sampling, is more extensively used in time-invariant systems as the theory of
uniform sampling (either instantaneous or otherwise) is well developed and the
techniques are easier to implement in practical systems.
Sampling Techniques:
There are three types of sampling techniques as under:
1. Ideal sampling or Instantaneous sampling or Impulse sampling

2. Natural sampling
3. Flat top sampling
1. Ideal sampling or Instantaneous sampling or Impulse sampling:

For the proof of sampling theorem we use ideal or impulse sampling.
The concept of ‘instantaneous’ sampling is more of a mathematical abstraction as no
practical sampling device can actually generate truly instantaneous samples (a
sampling pulse should have non-zero energy). However, this is not a deterrent in
using the theory of instantaneous sampling, as a fairly close approximation of
instantaneous sampling is sufficient for most practical systems. To contain our
discussion on Nyquist’s theorems, we will introduce some mathematical expressions.
If x(t) represents a continuous-time signal, the equivalent set of instantaneous uniform
samples {x(nTs)} may be represented as:
{x(nTs)} = Σ x(t).δ(t- nTs)
where x(nTs) = x(t) =nTs , δ(t) is a unit pulse singularity function and ‘n’ is an
integer

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Ideal sampling process
Figure 7(a)
Figure 7(b)

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2. Natural sampling:
In the analogue-to-digital conversion process an analogue waveform is sampled to

form a series of pulses whose amplitude is the amplitude of the sampled waveform at
the time the sample was taken. In natural sampling the pulse amplitude takes the
shape of the analogue waveform for the period of the sampling pulse as shown in
figure 8.
Figure 8
3. Flat Top sampling:
After an analogue waveform is sampled in the analogue-to-digital conversion process,

the continuous analogue waveform is converted into a series of pulses whose
amplitude is equal to the amplitude of the analogue signal at the start of the sampling
process. Since the sampled pulses have uniform amplitude, the process is called flat
top sampling as shown in figure 9.
Figure 9

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Note that due to the flat-top pulses, the spectrum of the sampled signal is distorted.
The narrower the pulse width lesser the distortion.
The original signal may be obtained by using a low-pass filter with a characteristic
which inverts the distortion.
Another important process in the PCM process is known as Quantization.
Sample & Hold circuit:
In electronics, a sample and hold circuit is used to interface real-world signals, by
changing analogue signals to a subsequent system. The purpose of this circuit is to
hold the analogue value steady for a short time while the converter or other following
system performs some operation that takes a little time.
Sampling mode:
In this mode, the switch is in the closed position and the capacitor charges to the
instantaneous input voltage.
Hold mode:
In this mode, the switch is in the open position. The capacitor is now disconnected
from the input. As there is no path for the capacitor to discharge, it will hold the
voltage on it just before opening the switch. The capacitor will hold this voltage till
the next sampling instant.
Sample and Hold Waveform

Figure 10
Now, from figure 10 the area under the curve (which is equivalent to the signal
power) is greater and so the filter output amplitude and quality of reproduced signal is
improved.
In most circuits, a capacitor is used to store the analogue voltage and an electronic
switch or gate is used to alternately connect and disconnect the capacitor from the
analogue input. The rate at which this switch is operated is the sampling rate of the
system.
In a sample and hold circuit the switch opens for a very short duration. The sample
and hold circuit integrates for a short duration charge into a capacitor.

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The 'hold' facility can be provided by a capacitor, when the switch connects the
capacitor to PAM output it charges to the instantaneous value.
A buffered sample and hold circuit consists of unit gain buffer preceding and
succeeding the charging capacitor. The high input impedance of the preceding buffer
prevents the loading of the message source and also ensures that the capacitor charges
by a constant rate irrespective of the source impedance see figure 11(a).
Sample Hold Circuit

Figure 11(a)
The high input impedance of the succeeding buffer prevents the charging from the
capacitor due to loading and hence the capacitor can hold the charge for infinite time,
at least theoretically. However, small leakage current through the capacitor dielectric
into '+'ve input of second buffer is always present which causes gradual charge loss.
The rate of change of voltage with respect to time dv / dt is called as droop rate and is
important parameter in sample and Hold circuit design. The sample and hold
waveform is shown in figure 11(b).
Sample and hold wave form

Figure 11(b)

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Important Parameters of Sample & Hold Circuit
1. Aperture time:
The aperture time is defined as the delay time between the beginnings of the hold
command to the time the capacitor voltage ceases to follow the information signal.
Hence the hold value is different from the true sample value. The aperture time cannot
be reducing to zero because on application of finite time taken by a switch to close &
open on application of the hold signal. Therefore a small value of aperture time is
sought after.
Timing Diagram for Sample and Hold Circuit

Figure 12
2. Acquisition Time:
In sample mode, it takes finite time for the capacitor to charge to the information
signal value depending on the RC time constant. This is called as the acquisition time.
The acquisition time is dependent on the current flowing from the input buffer
through switch and hence on RC time constant. The maximum acquisition time occurs
when the capacitor voltage has to change by the full amplitude of the information
signal.
3. Droop Rate:
As it has been discussed earlier, the presence of leakage current through capacitor
dielectric to +ve input of succeeding buffer causes charge loss of capacitor. Hence the
voltage level at the output falls with in time. This rate of change of voltage with
respect to time dv/dt is known as droop rate. Over value of droop rate is desirable as
the circuit should be able to maintain the sample at a relatively constant level until the
next sample.

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4. Feed Through:
At high frequencies, the stray capacitance within the switch causes some of the input
signal to appear at the output during the hold state (switch open). The fraction of input
signal appearing at the output of sample and hold circuit is called feed through.
The sample and hold feature provides both problem and benefit will be seen
afterwards.
Quantization:
In quantization the levels are assigned a binary codeword. All sample values falling
between two quantization levels are considered to be located at the centre of the
quantization interval. In this manner the quantization process introduces a certain
amount of error or distortion into the signal samples. This error known as quantization
noise is minimized by establishing a large number of small quantization intervals. Of
course, as the number of quantization intervals increase, so must the number or bits
increase to uniquely identify the quantization intervals. For example, if an analogue
voltage level is to be converted to a digital system with 8 discrete levels or
quantization steps three bits are required. In the ITU-T version there are 256
quantization steps, 128 positive and 128 negative, requiring 8 bits. A positive level is
represented by having bit 8 (MSB) at 0 and for a negative level the MSB is 1.
This is the process of setting the sample amplitude, which can be continuously
variable to a discrete value. Look at Uniform Quantization first, where the discrete
values are evenly spaced.
The operation of quantization is represented in figure 13. Here we contemplate a

signal m (t) whose excursion is confined to the range from VL to VH. We have divided
this total range into M equal intervals each of size S. Accordingly S, called the step
size, is S = (VH - VL)/M. In figure 13 we show the specific example in which M = 8. In
the center of each of these steps we locate quantization levels m0, m1,..., m7. The
quantized signal mq(t) is generated in the following way: Whenever m(t) is in the
range Δ0, the signal mq(t) maintains the constant level m0; whenever m(t) is in the
range Δ1, mq(t) maintains the constant level m1; and so on. Thus the signal mq(t) will
at all times be found at one of the levels mo, ml, ..., m7. The transition in mq (t) from
mq(t) = m0 to mq(t) = m1 is made abruptly when m(t) passes the transition level L01
which is midway between m0 and m1 and so on. To state the matter in an alternative
fashion, we say that, at every instant of time, mq(t) has the value of the quantization
level to which m(t) is closest. Thus the signal mq(t) does not change at all with time or
it makes a "quantum" jump of step size S. Note the disposition of the quantization
levels in the range from VL to VH. These levels are each separated by an amount S, but
the separation of the extremes VL and VH each from its nearest quantization level is
only S/2. Also, at every instant of time, the quantization error m (t) - mq(t) has a
magnitude which is equal to or less than S/2.

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Process of Quantization of an Analog Signal

Figure 13
We see, therefore, that the quantized signal is an approximation to the original signal.
The quality of the approximation may be improved by reducing the size of the steps,
thereby increasing the number of allowable levels. Eventually, with small enough
steps, the human ear or the eye will not be able to distinguish the original from the
quantized signal. To give the reader an idea of the number of quantization levels
required in a practical system, we note that 256 levels can be used to obtain the
quality of commercial color TV, while 64 levels gives only fairly good color TV
performance. These results are also found to be valid when quantizing voice.
Uniform Quantization
We assume that the amplitude of the signal m(t) is confined to the range (-mp, +mp ).
This range (2mp) is divided into L levels, each of step size δ, given by
δ = 2 mp / L
A sample amplitude value is approximated by the midpoint of the interval in which it
lies. The input/output characteristic of a uniform quantizer is shown figure 14.

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Figure 14
The conventional, practical digital-to-analog converter (DAC) does not output a
sequence of impulses (such that, if ideally low-pass filtered, result in the original
signal before sampling) but instead output a sequence of piecewise constant values or
rectangular pulses. This means that there is an inherent effect of the zero-order hold
on the effective frequency response of the DAC resulting in a mild roll-off of gain at
the higher frequencies (a 3.9224 dB loss at the Nyquist frequency). This zero-order
hold effect is a consequence of the hold action of the DAC and is not due to the
sample and hold that might precede a conventional ADC as is often misunderstood.
The DAC can also suffer errors from jitter, noise, slewing, and non-linear mapping of
input value to output voltage.
Each binary word defines a particular narrow range of amplitude level. The sampled
value is then approximated to the nearest amplitude level. The sample is then assigned
a code corresponding to the amplitude level, which is then transmitted.
This process is called as Quantization & it is generally carried out by the A/D
converter.

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Quantization & Encoding of a sampled signal
Figure 15
The PCM code generated after Quantization process:

010 101 111 111 110 010 001 010 010

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There are two important problems associated with quantization.

a. Quantization noise :
As we have seen the signal is approximated to the nearest level (step). Since the levels
are discrete where as the signal is continuous, the discrepancy creeps in.
The difference between the analog signal value & its approximated one (quantized
one) is random & unpredictable. This is a sort of unwanted, unpredictable, random
signal which accompanies the information signal and is termed as 'Quantization
noise'.
The difference between m (t) - mq(t) can be regarded as noise and is called
quantization noise. Hence, the received signal is not a perfect replica of the trans-
mitted signal m (t). The difference between them is due to errors caused by additive
noise and quantization noise.
Quantization Noise
Figure 16
Quantization noise can be reduced by increasing the number of levels, hence reducing
the approximation. But it can never be eliminated. Increasing the number of levels to
reduce quantization noise has the effect of increasing the number of bits. But nothing
comes without price. Increasing the number of bits to represent a sample increases the
system's bandwidth requirement.
b. Finite sampling time of A/D converter :
Another problem associated with quantization is that the A/D Converter requires
finite time to convert the analog information to digital data. The A/D Converter
requires that the value at its input, remain unchanged till the conversion is complete.
But in practice, the duration of sampled pulse is much smaller than the A/D
converter's sampling time.

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Bit step Analog to digital converter

Figure 17
This problem can be overcome by using a sample & hold circuit prior to A/D
converter output. The sample & hold circuitry holds the sample value till the next
sample. The encoding method described above is called as uniform encoding i.e. the
quantization levels are uniform for all the amplitude range. But this method of
encoding has disadvantages of its own. The quantization noise plays havoc with the
low level signals because the % approximation compared to the signal amplitude is
very high. This causes a great amount of distortion at the receiver for low level
signals. Also the quieter part of music or speech could become severely distorted &
would make them unpleasant to listen.
To overcome this problem, a non-uniform encoding scheme is used. Here the
quantization levels are clear together for low level than they are for the high levels.
This has an effect of compression on the extreme ends of the signal. The input/output
characteristics for compression signal passed through a comparator network 'prior to
compression (See figure 18). This process is called compression.

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An input output characteristic providing compression

Figure 18
The opposite effect is utilized at the receiver to undo the effect of compression, is
termed as expanding. The two processes are combined are known as compounding
this feature is not provided on trainer but you should be aware of its existence. Some
error correcting codes & synchronization can also be transmitted along with the
information signal.
At receiver, the data is decoded by the D/A converter; the recovered samples are
filtered & reconstructed to provide the original waveform.
Various channels can be multiplexed in time domain i.e. the information data from
various sources are sequentially transmitted over the same transmission medium e.g.
Let us assume a 3 channel PCM system. The system samples 0-2 samples sequentially
providing 3 samples to be converted to 3 "n" bit words. These three n bit words forms
the basis of a frame. The frame contains these three n bit words also contains some
synchronization & reference positioning information.
On more complex multi-channel systems, control & routing information have to be
included. This information is termed as signaling information. If all these information
can not be fitted in a single frame, a separate channel is used for signaling &
synchronization information.
In Europe, a 30 channel PCM System is followed which is specified by CCITT
(International Radio Consultative Committee). Besides these channels, two separate
channels are used for signaling & synchronization information. Here the multi frame
consists of 16 frames.
Multi Frame:
When the number of bits in allocated channels is insufficient to cope with the
synchronization & signaling information then it is spread on defined channels over a
number of frames. This sequence of frames is known as a Multi Frames.

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Pulse Code Modulation:

A signal, which is to be quantized prior to transmission, is usually sampled first. The
quantization is used to reduce the effects of noise, and the sampling allows us to time-
division multiplex a number of messages if we choose to do so. The combined
operations of sampling and quantizing generate a quantized PAM waveform, that is, a
train of pulses whose amplitudes are restricted to a number of discrete magnitudes.
We may, if we choose, transmit these quantized sample values directly. Alternatively
we may represent each quantized level by a code number and transmit the code
number rather than the sample value itself. The merit of so doing will be developed in
the subsequent discussion. Most frequently the code number is converted, before
transmission, into its representation in binary arithmetic, i.e., base-2 arithmetic. The
digits of the binary representation of the code number are transmitted as pulses.
Hence the system of transmission is called (binary) pulse code modulation (PCM).
We review briefly some elementary points about binary arithmetic. The binary system
uses only two digits, 0 and 1. An arbitrary number N is represented by the sequence...
k2 k1I k0, in which the k's are determined from the equation
N =. + k222 + k121 + k020
with the added constraint that each k has the value 0 or 1. The binary representations
of the decimal numbers 0 to 15 are given in table 1. Observe that to represent the four
Binary Codes for 16 Level Quantization
Table 1
(Decimal) numbers 0 to 3; we need only two binary digits k1 and k0. For the eight
(decimal) numbers from 0 to 7 we require only three binary places, and so on. In
general, if M numbers 0, 1, M - 1 are to be represented, then an N binary digit
sequence kN-1 ... k0 is required, where M = 2N.

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Sampling and Binary Coding of an Analog Signal (PCM)

Figure 19
The essential features of binary PCM are shown in figure 19. We assume that the
analog message signal m (t) is limited in its excursions to the range from - 4 to +4
volts. We have set the step size between quantization levels at 1 volt. Eight
quantization levels are employed, and these are located at - 3.5, - 2.5, +3.5volts. We
assign the code number 0 to the level at - 3.5volts, the code number 1 to the level at -
2.5volts, etc., until the level at + 3.5volts, which is assigned the code number 7. Each
code number has its representation in binary arithmetic ranging from 000 for code
number 0 to 111 for code number 7.
In figure 19, in correspondence with each sample, we specify the sample value, the
nearest quantization level, and the code number and its binary representation. If we
were transmitting the analog signal, we would transmit the sample values 1.3, 3.6,
2.3, etc. If we were transmitting the quantized signal, we would transmit the
quantized sample values 1.5, 3.5, 2.5, etc. In binary PCM we transmit the binary
representations 101, Ill, 110, etc.

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Electrical pulses for Binary Codes in PCM

Figure 20
We may represent the binary digits by electrical pulses in order to transmit the code
representations of each quantized level over a communication channel. Such a
representation is shown in figure 20. Pulse time slots are indicated at the top of the
figure, and, as shown in figure 20 a, a pulse represents the binary digit 1, while the
binary digit 0 is represented by the absence of a pulse. The row of three-digit binary
numbers given in figure 20 is the binary representation of the sequence of quantized
samples in figure 19. Hence the pulse pattern in figure 20a is the (binary) PCM
waveform that would be transmitted to convey to the receiver the sequence of
quantized samples of the message signal m (t) in figure 19. Each three-digit binary
number that specifies a quantized sample value is called a word. The spaces between
words allow for the multiplexing of other messages.
At the receiver, in order to reconstruct the quantized signal, all that is required is that
a determination be made, within each pulse time slot, about whether a pulse is present
or absent. The exact amplitude of the pulse is not important. There is an advantage in
making the pulse width as wide as possible since the pulse energy is thereby
increased and it becomes easier to recognize a pulse against the background noise.
Suppose then that we eliminate the guard time τg between pulses. We would then
have the waveform shown in figure 20 b.
We would be rather hard put to describe this waveform as either a sequence of
positive pulses or of negative pulses. The waveform consists now of a sequence of
transitions between two levels. When the waveform occupies the lower level in a
particular time slot, a binary 0 is represented, while the upper voltage level represents
a binary 1.Suppose that the voltage difference of 2 V volts between the levels of the
waveform of figure 20 b is adequate to allow reliable determination at the receiver of
which digit is being transmitted. We might then arrange, say, that the waveform make
excursions between 0 and 2 V volts or between - V volts and + V volts. The former
waveform will have a dc component, the latter waveform will not. Since the dc
component wastes power and contributes nothing to the reliability of transmission,
the latter alternative is preferred and is indicated in figure 20 b.

ST2113
The PCM System:

The Encoder:
A PCM communication system is represented in figure 21. The analog signal m (t) is
sampled, and these samples are subjected to the operation of quantization. The
quantized samples are applied to an encoder. The encoder responds to each such
sample by the generation of a unique and identifiable binary pulse (or binary level)
pattern. In the example of figures 19 and 20, the pulse pattern happens to have a
numerical significance that is same as the order assigned to the quantized levels.
However, this feature is not essential. We could have assigned any pulse pattern to
any level. At the receiver, however, we must be able to identify the level from the
pulse pattern. Hence it is clear that not only does the encoder number the level, it also
assigns to it an identification code.
A PCM Communication System

Figure 21
The combination of the quantizer and encoder in the gray outlined box of figure 21 is
called an analog-to-digital converter, usually abbreviated A/D converter. In
commercially available A/D converters there is normally no sharp distinction
between that portion of the electronic circuitry used to do the quantizing and that
portion used to accomplish the encoding. In summary, then, the A/D converter
accepts an analog signal and replaces it with a succession of code symbols, each
symbol consisting of a train of pulses in which each pulse may be interpreted as the
representation of a digit in an arithmetic system. Thus the signal transmitted over the
communications channel in a PCM system is referred to as a digitally encoded signal.
The Decoder:
When the digitally encoded signal arrives at the receiver (or repeater), the first
operation to be performed is the separation of the signal from the noise, which has
been added during the transmission along the channel. As noted previously,
separation of the signal from the noise is possible because of the quantization of the
signal. Such an operation is again an operation of requantization; hence the first
block in the receiver in figure 21 is termed a quantizer. A feature which eases the

ST2113
burden on this quantizer is that for each pulse interval it has only to make the
relatively simple decision of whether a pulse has or has not been received or which of
two voltage levels has occurred. Suppose the quantized sample pulses had been
transmitted instead, rather than the binary-encoded codes for such samples. Then this
quantizer would have had to yield, in each pulse interval, not a simple yes or no
decision, but rather a more complicated determination about which of the many
possible levels had been received. In the example of figure 21, if a quantized PAM
signal had been transmitted, the receiver quantizer would have to decide which of the
levels 0 to 7 was transmitted, while with a binary PCM signal the quantizer need only
distinguish between two possible levels. The relative reliability of the yes or no
decision in PCM over the multi valued decision required for quantized PAM
constitutes an important advantage for PCM.
The receiver quantizer then, in each pulse slot, makes an educated and sophisticated
estimate and then decides whether a positive pulse or a negative pulse was received
and transmits its decisions, in the form of a reconstituted or regenerated pulse train, to
the decoder. (If repeater operation is intended, the regenerated pulse train is simply
raised in level and sent along the next section of the transmission channel.) The
decoder, also called a digital-to-analog (D/A) converter, performs the inverse
operation of the encoder. The decoder output is the sequence of quantized multilevel
sample pulses. The quantized PAM signal is now reconstituted. It is then filtered to
reject any frequency components lying outside of the base band. The final output
signal m'(t) is identical with the input m (t) except for quantization noise and the
occasional error in yes-no decision making at the receiver due to the presence of
channel noise.
In telephony, a standard audio signal for a single phone call is encoded as 8000
analog samples per second, of 8 bits each, giving a 64k bit/s digital signal known as
DS0. The default signal compression encoding on a DS0 is either μ-law (mu-law)
PCM (North America and Japan) or a-law PCM (Europe and most of the rest of the
world). These are logarithmic compression systems where a 12 or 13 bit linear PCM
sample number is mapped into an 8-bit value. This system is described by
international standard G.711.
Differential (or Delta) pulse-code modulation (DPCM) encodes the PCM values as
differences between the current and the previous value. For audio this type of
encoding reduces the number of bits required per sample by about 25% compared to
PCM. Adaptive DPCM (ADPCM) is a variant of DPCM that varies the size of the
quantization step, to allow further reduction of the required bandwidth for a given
signal-to-noise ratio. Where circuit costs are high and loss of voice quality is
acceptable, it sometimes makes sense to compress the voice signal even further. An
ADPCM algorithm is used to map a series of 8 bit PCM samples into a series of 4 bit
ADPCM samples. In this way, the capacity of the line is doubled. The technique is
detailed in the G.726 standard. Later it was found that even further compression was
possible and additional standards were published. Some of these international
standards describe systems and ideas that are covered by privately owned patents and
thus use of these standards requires payments to the patent holders. Some ADPCM
techniques are used in Voice over IP communications.

ST2113
Differential Pulse-Code Modulation

In Practical system bandwidth requirement for transformation of information is very
important aspect, since if bandwidth requirement is less, more number of channels can
be multiplexed on a single line and full utility of transmitting media is extracted out.
In a system in which a base band signal m (t) is transmitted by sampling, there is
available a scheme of transmission which is an alternative to transmitting the sample
values (quantized or not) at each sampling time. We can instead, at each sampling
time, say the kth sampling time, transmit the difference between the sample value m(k)
at sampling time k and the sample value m(k - 1) at time k - 1. If such changes are
transmitted, then simply by adding up (accumulating) these changes we shall generate
at the receiver a waveform identical in form to m (t). There can be a difference in dc
components between transmitted and received signals but, almost invariably; such dc
components are of no interest.
Such a differential scheme has special merit when these differences are to be
transmitted by pulse code modulation. For we may well anticipate that the differences
m(k) - m(k - 1) will be smaller than the sample values themselves. Hence fewer levels
will be required to quantize the difference than are required to quantize m (k) and
correspondingly, fewer bits will be needed to encode the levels. For example,
suppose that m (k) extends over a range VH - VL, and using PCM, m (k) is encoded
using 28 = 256 levels. Then the step size is S = (VH - VL)/28, that is VH- VL = 256S. If,
however, the difference signal m (k) - m (k - 1) extends only over the range ± 2S then
the quantized levels needed are at ± O.5S and at ± l.5S. There are now only four
levels and two bits per sample difference are adequate.
In an analog system, where we are able, at least in principle, to transmit the
differences exactly, the differential system described above would operate in
accordance with our description. In a digital (quantized) system we encounter the
complication that the differences are not generally transmitted exactly because of the
quantization. Further, we have the problem that the difference may be larger than the
maximum that can be accommodated because of the restricted number of encoding
bits we have provided. Hence it might well be that at some time there might be a
large discrepancy between the original signal m(t) and the signal m(t) generated at the
receiver by accumulation. Suppose that over a number of samplings, while m (t) is
increasing, the transmitted differences were too small so that m (k) had fallen
substantially short of keeping up with m (t). Suppose, further, that in the interval
sampling times k and k + 1, m (t) should decrease slightly. Clearly if we transmitted
the negative change of m (t) we would be giving the wrong signal.
In a digital differential system we circumvent the difficulty we have just described by
making available at the transmitter a duplicate of the receiver accumulator so that at
the transmitter we have available the same signal m (t). Then we arrange that the
transmitted signal should not convey the most recent change in m (t) but conveys
instead the difference between m (t) and m’ (t).

ST2113
DPCM Transmitter / Receiver

Figure 22
In an analog system, this difference m (t) – m’ (t) is precisely the last change in m (t).
In a quantized system, as we have noted, such is not the case. In short, in a quantized
system we add or subtract from m (t) a value which is appropriate to bring m (t)
closer to m (t). The waveform m’ (t) is generally referred to as the approximation to
m (t).
Altogether, then, the quantized differential transmission scheme is as shown in 22
(we ignore initially the "predictors" that appear in the figure 22). The receiver
consists of an accumulator which adds up the received quantized differences ΔQ (k)
and a filter which smooths out the quantization noise. The output of the accumulator
is the signal approximation m’ (k) which becomes m (t) at the filter output. At the
transmitter we need to know whether m (t) is larger or smaller than m (t), and by how
much. We may then determine whether the next difference ΔQ (k) needs to be positive
or negative and of what amplitude in order to bring m’ (t) as close as possible to m
(t). For this reason we have a duplicate accumulator at the transmitter. At each
sampling time, the transmitter difference amplifier compares m’ (t) and m (t) and the

ST2113
sample and hold circuitry holds the result of that comparison Δ (t) for the duration of
the interval between sampling times. The quantized generates the signal So (t) = ΔQ
(k) both for transmission to the receiver and to provide input to the receiver
accumulator in the transmitter. In a practical system the quantized differences would
first be encoded into a binary bit stream before transmission and decoded at the
receiver. For simplicity the encoder and decoder are not included in the above figure
22.
It needs to be emphasized that the basic limitation of the scheme we have just
described is that the transmitted differences are quantized and are of limited
maximum value. The quantization means that almost never will the increment ΔQ (k)
added to m (k) make m’ (t) precisely equal to m (t). The limitation on the maximum
value of (k) means that when m (t) changes monotonically at a rapid rate, m’ (t) may
simply not be able to keep up.
Need for a Predictor:
The DPCM scheme we have described in this section turns out, as a matter of
practice, not to be effective. When the sampling rate is set at the Nyquist rate it
generates unacceptably excessive quantization noise in comparison to PCM. The
quantization noise can be reduced by significantly increasing the sampling rate. With
increased rate the differences from sample to sample are smaller and the rate of
producing large quantization errors is reduced. Suppose, then, that in DPCM we
increase the sampling rate, above the Nyquist rate, to the point where we get a quality
of sound transmission, which is comparable to that, available from PCM operating at
the Nyquist rate. Then again it turns out that DPCM is at a disadvantage because it
has been demonstrated that the bit rate of DPCM (bits per sample x sample rate)
exceeds that required for PCM.
The situation in DPCM can be improved by recognizing that there is a correlation
between successive samples of the signal m (t) and of Δ (t) if the signal is sampled at
a rate exceeding the Nyquist rate. Hence a knowledge of past sample values or
differences allows us to predict, with some probability of being correct, the range of
the next required increment. To take advantage of this correlation, a predictor is
included in the DPCM system shown in figure 22 the predictor will generally be a
moderately sophisticated system; it will need to incorporate the facility for storing
past differences and for carrying out some algorithm to predict the next required
increment.
Altogether, the quality of voice or video transmission using DPCM can be made
comparable to that of PCM by increasing the sampling rate (which reduces the
differences and increases the correlation between samples) and by using a predictor.
Most importantly, by these expedients, DPCM can operate at approximately one-half
of the bit rate of PCM with a consequent saving of spectrum space.

ST2113
Practical Considerations:
In PCM system we use 8 bits per samples, so if 8 KHz is the sampling frequency then
64K bits/s is the bandwidth requirement. If we can reduce number of bite per sample
and keep sampling frequency same, it will effectively reduce bit rate of the system
and hence the bandwidth required for the transmission. In DPCM we transmit the
difference between previous sample and current sample instead of transmitting exact
sample value every time. Naturally the difference between the two consecutive
samples is less than the sample value itself. So it requires less numbers of bits to
quantize the difference signal.
In ST2113 we used 4 bits for quantizing the difference and 1 bit for positive or
negative difference indication. So, overall 5 bits per sample are transmitted.
Sampling frequency is fixed and is 8 KHz. Both transmitter and receiver are designed
on the same board. In actual practical system we have to duplicate receiver at
transmitting end itself. This receiver consists of an accumulator and a predictor with
associated computing circuitry. The output of this receiver is the predicted output
computed with the help of the accumulator. Now this predicted output and the original
modulating signals are fed into the input terminals of the difference amplifiers. At the
O/P of subtractor (or difference amplifier) we get the difference between the previous
sample and current value of the signal at the sampling instance. This can be positive
or negative depending upon their relative amplitudes. A sign bit is generated
accordingly using a comparator. If difference is positive i.e. current value is more than
the predicted value a ‘0’ (or low signal) is generated, and if the difference is negative
a high signal (or logic‘1’) is generated. This sign signal is also send together with the
quantized difference signal.
In either case input to A/D should be positive. For this purpose two channels of A/D
are used & giving the sign bit to the control input of multiplexer of ADC makes
selection of particular channel. The negative difference signal is first inverted to
positive and then applied to the second channel of the multiplexer. The first channel is
being fed with positive polarity difference signal directly. Now as per the nature of
difference signal i.e. positive or negative the input to the ADC is positive always but
the sign bit later on as to differentiate between the positive and negative sampled
values controls the rest of the processing.
Only four least significant bit values of the ADC are used for the conversion purpose
since the value of difference signal between present and previous sampled signal is
always less than the individual sampled value and hence only few numbers of bits are
suffice to quantize the difference. Rest of the MSBs are used to limit the output by
generating a control signal which will either keep on adding or subtracting the
samples difference until the difference between predicated and present samples comes
within the specified limits.
The outputs of ADC together with the sign bit are then passed to the parallel to serial
converter. After the serial conversion we are ready to transmit it through the
communication channel and to the input of duplicated receiver at the transmitting end
as well.

ST2113
At the receiver end all the received serial bits are fed to the serial to parallel converter.
After getting the parallel bits these are fed simultaneously to the adder and subtractor
block, which are the part of signal processing circuit. Activation of particular block is
decided by the nature of the sign bit. If the present sample value is less than the
previously stored sample the difference would be positive and thus resulting in a high
sign bit value. This high sign bit will activate the adder unit and the difference value
will get added to the previous sampled value and the process keep going like this until
the present sample value exceed the previous sample value. In that case the difference
would be negative and hence the subtractor unit gets activated thus causing the
present value to get subtracted from the previously stored sample value. The whole
process is such that it keeps tracking the input signal and thus the stored values
replicate the input signal in discrete fashion. The stored samples are then converted
into analog form through D to A converter. The output of this converter is then fed to
an output low pass filter and to the “predicted O/P” input of the DPCM transmitter.

ST2113
Brief Description of ST2113

ST2113, Differential pulse code modulation / Demodulation Trainer has been
designed to clearly demonstrate the complete processes involved in Differential Pulse
Code Modulation and demodulation technique. The detailed mimic is shown in the
figure 23, given below. ST2113, Differential Pulse Code Modulation / Demodulation
Trainer board consists of following major blocks:
(a) Audio signal generator: The audio signal generator block generates sine wave
and square wave output in the frequency range of 300Hz to 3.4 KHz i.e. in the
audio frequency range. The frequency of the output signal can be adjusted with
the help of pot provided therein. This audio signal generator serves the purpose
of test signal source for the modulator / demodulator.
(b) Clock and control signal block: This block consists of various control and
clock signal test points. These clock and control signals guide the various
processes in the system and provide a smooth and errorless flow of the data and
signals in the system. A timing diagram of all these control and clock signals is
given in the figure 24. This timing diagram should be taken as a reference while
performing the experiments.
(c) Differential pulse code modulator (or transmitter) : The modulator
comprises of :
i. Difference Amplifier
ii. Sign bit generator
iii. ADC
iv. Parallel to serial converter
v. Predictor

ST2113
ST2113 Differential Pulse Code Modulation / Demodulation Trainer Mimic

Figure 23

ST2113
(i) Difference amplifier: The difference amplifier subtracts the predicted

estimate of the input signal from the original input signal and then gives this
difference signal to the ADC for A/D conversion.
(ii) Sign Bit Generator: The sign bit generator generates the sign bit based on
the sign of difference signal sample. If the magnitude of the incoming signal
is larger than the predicted signal, the sign bit generator will generate a high
signal (logic ‘1’) or else it will generate low signal (logic ‘0’). This sign bit
controls the mathematical manipulation of data in the predictor accumulator.
(iii)ADC: The ADC converts the analog difference signal sample into digital
data bits. Note that the bits required to encode the difference signal would
obviously be fewer than had it been the encoding of complete input signal
sample (refer to the theory of DPCM technique given in the previous
section). In the present system only four bits are being used to encode the
difference signal. The main control signals associated with this A/D
conversion are R/W signal and output LE (latch enable) signal of the ADC.
The frequency of R/W control signal is 8 KHz, which can be termed as the
sampling frequency of the system.
(iv) Serial to Parallel converter: The serial to parallel converter converts
parallel data (four bits) coming from A/D converter and an extra sign bit into
the serial form. The main control and clock signals for S/P conversion are
EN signal and Clock of shift register (refer o the timing diagram given in
figure 24). Note that during the time period when enable signal is high,
exactly five clock pulses drives the S/P converter in order to convert five
parallel bits into serial form. The digital signal (or data) coming out of this
converter is the final differentially coded PCM signal available for
transmission.
(v) Predictor: The Predictor is nothing but a receiver used at the modulator end
itself. In fact the similar circuit, used here for predictor, is used at the
receiver end as a receiver with an added output filter. In actual practical
system two receiver configurations are used; one at the transmitting end as a
predictor accumulator and the other at receiving end as a receiver. The
filtered output of the predictor accumulator in the present system can serve
here the purpose of received signal and so there is no need of providing a
redundant section as far as the conceptual study of the DPCM technique is
concerned.
(d) Differential pulse code demodulator (or Receiver) : The Receiver consists of
the following sections :
i. Serial to Parallel converter
ii. Adder and Subtractor section
iii. Data Latches
iv. DAC
v. Output LPF

ST2113
(i) Serial to Parallel converter: The serial to parallel converter at the receiver
end converts the received data bits into parallel form. The clock signal used
here is same as used in P/S conversion at the transmitter. After S/P
conversion, out of five received bits, sign bit is sent to control circuitry and
rest four bits are sent to the signal processing section.
(ii) Adder and Subtractor section: This section does the main computational
task. The circuitry adds or subtracts the incoming coded difference signal
from the difference signal, which in this case is the output of sign bit
generator that has been extracted out through S/P conversion. Actually this
sign bit enables or disables the adder or subtractor block based on whether it
is ‘0’ or ‘1’ at that particular instant.
(iii)Data Latches: These latches act in dynamism to store and forward he
processed data to next sections.
(iv) DAC: DAC converts the digitally processed data back into analog form..
The data available at the output of the latches after addition or subtraction of
the differentially encoded data from the previously stored data goes to the
input of DAC.
(v) Output LPF: The output LPF filter the output obtained from the ADC and
thus reduces the quantization noise and smoothened the signal.
(e) Audio Input circuit: This is basically an audio amplifier which amplifies the
output of an audio input transducer (e.g. mic). The electrical form the audio then
can be used as a test signal to check the capability of the system for faithful
transmission of the voice signal through DPCM process.
(f) Audio output circuit: The audio output amplifier amplifies the output of LPF
and then this output can be fed to an audio transducer such as headphone to get
the audio output back.

ST2113
Timing Diagram of various Controls and Clock Signals

Figure 24

ST2113
Experiment 1
Objective: Study of Differential pulse code modulation and Demodulation
Technique
Equipments Required:
1. ST2113 DPCM Trainer with power supply cord
2. CRO with connecting cable
3. Connecting Cords
Connection Diagram:
Setup for study of DPCM Process

Figure 1.1

ST2113
Procedure:
1. Connect the sine wave output of the audio signal generator to one of the inputs
of the difference amplifier as shown in the figure 1.1.
2. Connect the output of the difference amplifier to the input of the ADC.
3. Connect the output of parallel to serial converter to the input of serial to
parallel converter of the receiver accumulator.
4. Connect the output of the DAC to the other input of the difference amplifier as
shown in the above figure 1.1.
5. Now switch ‘On’ the power supply. Observe the sine wave output of the sine
wave generator on the CRO. Adjust the frequency of sine wave at 1 KHz.
6. Observe the signal at the output of DAC. You will see the stair case
approximation of the input signal at the output of DAC. Also observe the
output of low pass filter and see that it is nearly same as the input signal.
7. Now observe various controls and clock signal shown in the control and clock
section and try to relate these signals with the timing diagram of figure 1.2.
8. Note that the sampling starts with the R/W signal pulse that has a frequency of
8 KHz. For a small duration when pulse is high, ADC reads the input port and
for the rest of the low period it provides this data at the output of the ADC.
9. Now observe the clock and reset signal of the parallel to serial converter.
Observe that for the time period when reset is low, exactly five clock pulses
shift the content of the shit register.
10. Observe the LE2 and OE 2of previous data latch and relate them with timing
diagram of figure 1.2.
11. You may find it difficult to appreciate the entire control signals
simultaneously with a normal two channels CRO but nevertheless an intuitive
sense of relative time based occurrences of all these signals can make the task
easier. You can see all these signals simultaneously on the screen with the help
of logic analyzer too.
12. Also it is to be taken into consideration that the data flow through the entire
system is fast enough random as well so it is not feasible to observe the data
exactly at any point. But still you can have an overview of the data statistics at
any test point of the data bus.

ST2113
Figure 1.2
Observations:
Conclusion:
Questions:
1. Define DPCM?
2. What are the limitations of PCM system?
3. What is the function of Predictor?
4. What is the function of differential amplifier?
5. Differentiate between PCM and DPCM?

ST2113
Experiment 2
Objective: To verify experimentally that DPCM is a differentiation process.
3. connecting Cords
Connection Diagram:
Setup for DPCM modulation /Demodulation for a square wave input

Figure 2.1

ST2113
Procedure:
1. Connect square wave output of the signal generator to one of the input of the
difference amplifier.
2. Make rest of the connections as per the procedure given in the previous
experiment. You can refer to the figure 2.1 while making connections.
3. Switch ‘On’ the power supply and Set the frequency of square wave around 1
KHz.
4. Connect the square wave output and the filtered output (after DAC) to the
CRO channels and see the waveform in dual trace mode.
5. Observe that corresponding to rising and falling edges of the square wave we
get positive and negative impulses in the demodulated waveform as shown in
the figure 2.2 given below.
Square wave input with demodulated output

Figure 2.2
6. The above figure 2.2 suggests that the DPCM process is nothing but a
differentiation process.
The above result is obvious from point of view of the process since at the
rising or falling edges, the difference between the original samples and
predicted samples would be very large but subsequently the at the constant
amplitudes of the input signal the circuit try to minimize the difference as to
track the input signal and correspondingly we get ramp shape for a certain
time period during this constant amplitude of the input signal. The slop of this
ramp will depend upon the system frequency (sampling frequency of the
system) i.e. higher the sampling rate; steeper will be the slop of ramp.
Observations:
Conclusion:

ST2113
Questions:
1. What is the bandwidth requirement for DPCM?
2. Draw the block diagram of DPCM?
3. What are the draw backs of DPCM?
4. What is slope overload?
5. How slope over loading can be reduced?

ST2113
Experiment 3
Objective: To establish voice link using DPCM Technique
3. Connecting Cords
Connection Diagram:
Setup for establishing voice link through DPCM technique

Figure 3.1

ST2113
Procedure:
1. Connect microphone jack to the input socket of the Audio I/P circuit.
2. Connect the O/P of the Audio I/P Circuit to one of the input of the difference
amplifier as shown in the figure 3.1.
3. Connect the output of difference amplifier to the ADC input.
4. Connect the output of parallel to serial converter to the serial parallel converter.
5. Connect the output of DAC to the other input of difference amplifier.
6. Connect the output of low pass filter to the input of Audio O/P circuit.
7. Connect the output socket of the Audio O/P circuit to the headphone/speaker.
8. Now switch ‘On’ the power supply.
9. Verify that voice link has been established. Also verify that the main
articulations of voice (or intelligibility of voice) are being properly carried out
through the system.
10. Use the microphone and the headphone supplied with the instrument only.
11. You can also test signals at various test point with a CRO prob.
12. The data, which has to be transmitted to the communication channel, is the
serial data present at the output of parallel to serial converter.
Observations:
Conclusion:
Questions:
1. Define the frequency band for voice?
2. What is the function of microphone?
3. Explain the voice communication process using DPCM technique?
4. Why low pass filter is used for reconstruction of signals?
5. What is the function of quantizer?

ST2113
Frequently Asked Questions

1. How PAM signal is converted into PCM signal?
Ans: A PAM system can be converted into a PCM system by adding a suitable
analogue-to-digital (A/D) converter at the source and a digital-to-analogue (D/A)
converter at the destination.
2. List the steps to get the PCM signals?
Ans: In PCM, modulation process is executed in three steps:
1. Sampling
2. Quantizing
3. Coding
3. What is Quantization?
Ans: In quantization the levels are assigned a binary codeword. All sample values
falling between two quantization levels are considered to be located at the centre of
the quantization interval. In this manner the quantization process introduces a certain
amount of error or distortion into the signal samples.
4. What is Quantization noise?
Ans: During the quantization process introduces a certain amount of error or
distortion into the signal samples. This error known as quantization noise
5. How Quantization noise can be minimized?
Ans: Quantization noise can be minimized by establishing a large number of small
quantization intervals. Of course, as the number of quantization intervals increase, so
must the number or bits increase to uniquely identify the quantization intervals.
6. Draw the block diagram of Bit step Analog to digital converter?
Ans: The block diagram of Bit step Analog to digital converter is as follows:

ST2113
7. List the pulse modulation techniques and explain them in short?

1. Pulse Amplitude Modulation (PAM) :
In pulse amplitude modulation system the amplitude of the pulse is varied in
accordance with the instantaneous level of the modulating signal. Now days, the
PAM system is not generally used, but it forms the first stage of the other types
of pulse modulation.
2. Pulse Width Modulation (PWM) :
In PWM system the width of the pulse is varied in accordance with the
instantaneous level of the modulating signal.
3. Pulse Position Modulation (PPM) :
In PPM System, the position of the pulse relative to the zero reference level is
varied in accordance with the instantaneous level of the modulating signal.
4. Pulse Code Modulation (PCM) :
In PCM System the amplitude of the sampled waveform at definite time
intervals is represented as a binary code. The first three techniques of the above
described systems are not truly digital but in fact are analog in nature. The very
fact that the variation of a particular pulse parameter is continuous rather than
being in the discrete steps makes the system analog in nature.
8. What are the benefits of using TDM?

Ans: Benefits of TDM are as follows:
1. TDM is all about cost: fewer wires and simpler receivers are used to transmit
data from multiple sources to multiple destinations.
2. TDM also uses less bandwidth than Frequency-Division Multiplexing (FDM)
signals, unless the bit rate is increased, which will subsequently increase the
necessary bandwidth of the transmission.
3. An asset of TDM is its flexibility. The scheme allows for variation in the
number of signals being sent along the line, and constantly adjusts the time
intervals to make optimum use of the available bandwidth. The Internet is a
classic example of a communications network in which the volume of traffic
can change drastically from hour to hour.
9. What is Pseudo-random Noise (PN) sequence?
Ans: A Pseudo-random Noise (PN) sequence is a sequence of binary numbers, e.g.
±1, which appears to be random; but is in fact perfectly deterministic. The sequence

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appears to be random in the sense that the binary values and groups or runs of the
same binary value occur in the sequence in the same proportion they would if the
sequence were being generated based on a fair "coin tossing" experiment.
10. What are the uses of Pseudo-random Noise (PN) sequence?
Ans: They can be used to logically isolate users on the same frequency channel. They
can also be used to perform scrambling as well as spreading and dispreading
functions.
11. Draw the block diagram and explain how PN sequence can be generated?
Ans: A PN generator is typically made of N cascaded flip-flop circuits and a specially
selected feedback arrangement as shown below.
Figure
The flip-flop circuits when used in the cascaded manner is called a shift register, since
each clock pulse applied to the flip-flops causes the contents of each flip-flop to be
shifted to the right. The feedback connections provide the input to the left-most flip-
flop. With N binary stages, the largest number of different patterns the shift register
can have is 2N. However, the all-binary-zero state is not allowed because it would
cause all remaining states of the shift register and its outputs to be binary zero. The
all-binary-ones state does not cause a similar problem of repeated binary ones
provided the number of flip-flops input to the module 2 adder is even. The period of
the PN sequence is therefore 2N-1, but IS-95 introduces an extra binary zero to
achieve a period of 2N, where N equals 15.
12. Write the advantages and disadvantages of Digital modulation system?
Ans: Advantages of digital modulation system:
a. Noise & Distortion :
a. Pulse which becomes distorted by the addition of noise can be reshaped at
the regenerators installed at pre-determined intervals along the link. Thus
within certain threshold the error will not creep in.
b. Multiplexing :
a. The information once sampled & coded can be multiplexed in time
domain, i.e. the coded information from different sources can be sent, one
after another, if it can be re-routed to the corresponding channels at the
receiver.
b. The information is coded in binary form, the source of information /
sample, becomes unimportant. Therefore many different sources such as

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telephone, facsimile, telegraphy and video cap are transmitted over same
channel & circuitry.
c. Store & forward (S & F) facility :
a. That information which has been binary coded in digital format can be
easily stored in the computer or memory elements, & information can be
forwarded at the desired time. It is required at the time of channel
congestion. The message can be stored in memory. Once the channel
becomes clear, the message can be forwarded to the called party.
d. Encryption & security :
a. The digital devices today are capable of high grade encryption. The data
can not be correctly interpreted if the receiver has no proper decoder.
Hence the digital communication can be highly secured.
e. Power requirement :
a. To transmit the digital data over the same channel requires less signal
power than that would be required for same performance of the receiver
for analog systems.
Disadvantages of digital modulation communication system:
a. Band with requirement :
The digital communication systems need very large bandwidth as compared to
its analog counter part.
b. Complexity :
The digital transmitter & receivers is the complex due to the requirement of
highly reliable timing information. This adds to complexity as well as to the
cost of the communications system. With the advent of new technology, the
digital circuits / IC's are becoming more and cheaper still prices are slightly at
the higher side. But the advantage offered by the digital techniques far over
weighs this consideration.
13. How many methods are there to transmit the data from one place to other?
Ans: There are two methods for sending digital data over a distance, namely
a. Parallel transmission
b. Serial transmission
14. Describe the use of serial and parallel transmission?
Ans: In short distance communication like inside terminal equipment or two
computer terminals located near each other, the signals are passed in parallel, format
over parallel wires. Thus the signal in the form of a word is passed. This mode is
faster.
For long distances, even more than few feet’s, this is uneconomical & inefficient way
of transmission. It is a wasteful of transmission media as each bit requires a separate
link. Therefore the digital signals are transmitted serially over a single link.

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15. What do you mean by sampling?

Ans: To convert continuous time signal to discrete time signal, a process is used
called as sampling.
16. What is sampling theorem?
Ans: The Sampling Theorem states that a signal can be exactly reproduced if it is
sampled at a frequency Fs, where Fs is greater than twice the maximum frequency
Fmax in the signal.
Fs > 2· Fmax
17. What is Nyquist frequency?
Ans: The frequency 2· Fmax is called the Nyquist sampling rate. Half of this value,
Fmax, is sometimes called the Nyquist frequency.
18. List different sampling techniques?
Ans: There are three types of sampling, which are as follows:
1. Ideal sampling or Instantaneous sampling or Impulse sampling
2. Natural sampling
3. Flat top sampling
19. What is under sampling?
Ans: When the sampling rate is lower than or equal to the Nyquist rate, a condition
defined as under sampling, it is impossible to rebuild the original signal according to
the sampling theorem.
20. What do you mean by aliasing?
Ans: Aliasing is the presence of unwanted components in the reconstructed signal.
These components were not present when the original signal was sampled. In
addition, some of the frequencies in the original signal may be lost in the
reconstructed signal. Aliasing occurs because signal frequencies can overlap if the
sampling frequency is too low. As a result, the higher frequency components roll into
the reconstructed signal and cause distortion of the signal Frequencies "fold" around
half the sampling frequency. This type of signal distortion is called aliasing.
21. Explain the process of sample and hold?
Ans: In electronics, a sample and hold circuit is used to interface real-world signals,
by changing analogue signals to a subsequent system. The purpose of this circuit is to
hold the analogue value steady for a short time while the converter or other following
system performs some operation that takes a little time.
Sampling mode:
In this mode, the switch is in the closed position and the capacitor charges to the
instantaneous input voltage.
Hold mode:
In this mode, the switch is in the open position. The capacitor is now disconnected
from the input. As there is no path for the capacitor to discharge, it will hold the

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voltage on it just before opening the switch. The capacitor will hold this voltage till
the next sampling instant.
22. How aliasing is removed?
Ans: Aliasing is removed by simply filtering out all the high frequency components
before sampling.
23. List methods to avoid aliasing?
Ans: To avoid the aliasing there are two approaches:
1. To raise the sampling frequency to satisfy the sampling theorem,
2. The other is to filter off the unnecessary high-frequency component from the
continuous-time signal. We limit the signal frequency by an effective low pass filter,
called anti aliasing pre filter, so that the remained highest frequency is less than half
of the intended sampling rate. If the filter is not perfect we must give some allowance.
24. What are active and passive filter?
Ans: filter is a network designed to pass signals having frequencies within certain
bands (called pass bands) with little attenuation, but greatly attenuates signals within
other bands (called attenuation bands or stop bands).
A filter network containing no source of power is termed passive, and one containing
one or more power sources is known as an active filter network.

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Warranty
1. We guarantee this product against all manufacturing defects for 24 months from
the date of sale by us or through our dealers. Consumables like dry cell etc. are
not covered under warranty.
2. The guarantee will become void, if
a) The product is not operated as per the instruction given in the Learning
Material
b) The agreed payment terms and other conditions of sale are not followed.
c) The customer resells the instrument to another party.
d) Any attempt is made to service and modify the instrument.
3. The non-working of the product is to be communicated to us immediately giving
full details of the complaints and defects noticed specifically mentioning the
type, serial number of the product and date of purchase etc.
4. The repair work will be carried out, provided the product is dispatched securely
packed and insured. The transportation charges shall be borne by the customer.
List of Contents
1. Mains Cord ............................................................................................... 1 No.
2. 2MM Patch Cords (16”). ........................................................................ 5 Nos.
3. Microphone .............................................................................................. 1 No.
4. Headphone ................................................................................................ 1 No.
5. Learning Material (CD) ............................................................................ 1 No.

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Differential Pulse Code

An ISO 9001:2008 company

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Differential Pulse Code Modulation Trainer

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For your safety:

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PCM block Diagram

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Pulse Code Modulation

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Basic Sampling Process

Sampling of signal at sampling interval (period) Ts

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Sampled Output of narrow pulses δ (t)

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Nyquist Criterion (Sampling Theorem)

For audio signals the highest frequency component is 3.4 KHz.

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Nyquist’s Uniform Sampling Theorem for Low pass Signal:

There are three types of sampling techniques as under:

1. Ideal sampling or Instantaneous sampling or Impulse sampling

1. Ideal sampling or Instantaneous sampling or Impulse sampling:

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Ideal sampling process

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In the analogue-to-digital conversion process an analogue waveform is sampled to

3. Flat Top sampling:

After an analogue waveform is sampled in the analogue-to-digital conversion process,

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Sample and Hold Waveform

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Sample Hold Circuit

Sample and hold wave form

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Important Parameters of Sample & Hold Circuit

Timing Diagram for Sample and Hold Circuit

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The operation of quantization is represented in figure 13. Here we contemplate a

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Process of Quantization of an Analog Signal

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Quantization & Encoding of a sampled signal

The PCM code generated after Quantization process:

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There are two important problems associated with quantization.

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Bit step Analog to digital converter

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An input output characteristic providing compression

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Pulse Code Modulation:

Binary Codes for 16 Level Quantization

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Sampling and Binary Coding of an Analog Signal (PCM)

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Electrical pulses for Binary Codes in PCM