Lab Manual
Lab Manual
Lab Manual
LABORATORY MANUAL
SYLLABUS
Digital Communication Lab (EC 591)
Contact: 3P
Credit: 2
(Students are required to perform at least ten experiments taking 3 from Group A, 3 from
Group B, 3 from Group C and 1 from Group D)
GROUP - A (At least three experiments)
1. Design, implementation and studies of the properties of 15 bit P.N. Sequence using shift register
2. Studies of the properties of A/D and D/A converter (AD7820/ADC0820 and ICL 8018A/8019A/8020A).
(Properties like transfer characteristics, code central line method of nonlinearity study, differential
nonlinearity, integral nonlinearity, resolution etc).
3. Study of pulse amplitude modulation and demodulation.
(Studies of distortion factor of the constructed signal as a function of signal frequency & amplitude. Further
study of distortion factor of filtered reconstructed signal as a function of sampling frequency and thus verify
the sampling theorem).
4. Studies of PCM transmitter and receiver.
(To measure the bit rate, bandwidth requirement and distortion factor of the reconstructed signal in presence
of channel noise).
5. Study of line coders: UPNRZ, PRZ, BPRZ, PNRZ
(To study the nature of waveform in CRO and its spectrum by spectrum analyzer. At least any one of the
line coders has to design, fabricated and tested).
6. Studies of Manchester coding and decoding technique.
(Studies of the nature of waveform, spectrum and self-synchronizing characteristic).
GROUP - B (At least three experiments)
7. Studies of PSK modulator and demodulator, connected by physical channel.
8. Studies on FSK modulator and demodulator, connected by simulated channel
9. Studies on ASK modulator and demodulator, connected by physical channel
10. Studies on QPSK modulator and demodulator, connected by either physical or simulated channel (In all
above experiments, nature of the modulated waveform is to be studied by a CRO. The spectrum is to be
studied with a spectrum analyzer and the essential bandwidth is to be determined; finally the reception
quality is to be analyzed by cross co-relation characteristics and measurement of bit error rate in presence of
channel noise).
Experiment No. – 1
OBJECTIVE: This experiment will enable one to understand the pseudo noise(pn)
sequence with certain autocorrelation properties.
THEORY : In spread spectrum system, the signal spreading code is the so called the
pseudo noise(pn) sequence which is generally periodic and consists of a periodic
coded sequence of 1’s and 0’s with certain autocorrelation properties. These signals
are pseudo random is as much as they appear to be unpredictable to be an outsider,
though they appear or can be generated by deterministic means by the person for
whom they are intended. When a shift resister has a non-zero initial state and the
output is fed back to the input , the unit acts as periodic sequence generation. In
general the longest possible sequence from a resister n stages is N= (2n – 1).The
corresponding output is called a maximum length sequence or PN sequence.
The names pseudo noise comes from the correlation properties of sequence of N, is
very large and Tb is very small, then the pn signal acts essentially like white noise with
a small DC component and hence is called Pseudo Noise Sequence.
RESULTS:
APPARATUS USED:
SL. ITEM
NO.
1 Bread Board
2 CRO
3 Connecting Wires
4 IC 7495, IC 7486
Experiment No: 2
OBJECTIVE: This experiment will enable one to sense that to every input digital
word a single output analog value corresponds.
THEORY:
The digital to analog converters compose the devices transforming a digital word
binary encoded and generated for example by a computer, into a discrete analog
signal, in the sense that to every input digital word a single output analog value
corresponds.
The ideal characteristic of a 3 bit DAC is shown in Fig.1 and it is represented by the
straight line that connects the output discrete values corresponding to the single input
digital data
FS
7.5
Analog
Output
Q
Let's consider now the possibility where only the most significant bit (MSB)
7
S3 is at level in the R resistance the current 13 = V REF / R will flow and therefore at
the operational amplifier output we will have the voltage
V 03 = -13 (R / 2) = -V REF /2
Analogously the contribution to output voltage provided by the immediately less
significant digit will result
V 02 = -12 (R / 2) = -(VREF / 2R) (R /2) = -VREF/ 4
and so on
Vo1=-II(R/2) -(VREF 14R) (R 12) = -VREFI 8
V 00 = 10 (R / 2) = -(VREF / 8R) (R / 2) = -VREF / 16
Note: In ST2602 the switches S3 -So are carried out with mechanical switches only
for simplicity and for educational reasons; in reality the integrated DAC use static
switches carried out in bipolar or CMOS technology.
D/ A converter with ladder network
The weighted resistances converter has been among the first ones studied and carried
out but it has been then abandoned in favor of the configuration with ladder network
that uses only resistances of two values, one the double of the other. The converter
with ladder network (R -2R ladder) eliminates the disadvantages of the weighted
resistances converter since it is characterized by the presence of only two types of
resistances, even if the resistance number, with bit equality, now results double, as it
is shown in Fig.3 for the 4 bit possibility, where the switches S3 -So connect the
corresponding resistance to the reference voltage V REF or to ground in accordance
with the fact that the corresponding bit finds itself to logic level 1or o.
If we suppose at first that all the input bits are at low level (S 3 S2 Sl So = 0000)
every switch grounds the respective resistance 2R; in these conditions, since point V
is a virtual ground, the nodes A,B,C,D see toward ground, and in every direction,
always the same resistance 2R. Let's consider now the possibility where only the most
significant bit S3 is at level 1: the current 13, delivered from the reference voltage V
REF, will divide in node A into two identical currents but equal to the half of the
incoming one, as it is shown in Fig.4
Being
I3 = VREF/2R+(2R//2R) = VREF/3R
the potential of node A against ground is equal to
VA3 =2R. 13/2 = (1/3).VREF
and therefore at the operational amplifier output the voltage results
V03 = -3 R/ 2R. (1/3) .VREF = -V REF/2
Analogously the contribution to the output voltage of the immediately less significant
digit is determined by noticing that the current delivered from the reference voltage V
REF is equal to
12 = VREF/2R+(2R//2R) = VREF/3R = I3
and it divides itself in node B into two identical currents of value 12/2: this now
current will split equally at node A so the potential of node A against ground is now
equal to
VA2 = 2R. (12/4) = (1 16) .VREF
and therefore at the operational amplifier output the voltage results
V02 = -3R 12R. 1/6 VREF = -VREF /4
V00 = -V REF / 16
By applying the principle of effect superposition the output voltage results at the end
Vo = -V REF (8.S3 + 4. S2 + 2.S1 + I.So) /16
When all the bits are at logic level 1 the output voltage assumes the maximum full
scale value
VOFS = -0.9375 VREF
While the quantum that represents the increase of the output voltage in
correspondence of the least significant bit (LSB), results
Q = -0.0625 V REF
Analog to digital
THEORY:
The digital systems have been spreading considerably since they allow realizing
Complex functions accurately, low cost and in a way comparatively simple to
implement. Analog/digital converters are the natural interface between the world of
physical quantities that vary analogically i.e. in a continuous way and that of the
digital control systems which vary with finite increases. AID converters have
numerous applications and consequently have a high diffusion. Just think for example
of the digital multi-meters. Each one of this contains an AID converter for converting
the analog quantity that has to be measured to corresponding numerical value.
10
Q
111
110
101
100
011
010
001
000
1.25 2.5 3.75 5 6.25 7.5 8.75
The straight line is the continuous analog signal, while the staircase line is the
quantized corresponding signal. If we assume a binary code, the 8 quantized states are
coded through a 3 bits digital word and this corresponds to the output of a 3 bits AID
converter. The sequences of binary numbers starting from "000" and reaching "111"
are assigned to the 8 output states. Let us anal# in detail a few aspects of the quantized
Signal. The first aspect is the “ution'" defined as the number of output states that can
be coded through a binary word of n bits; with n bits we can code 2 n output states. In
this case we have a 3 bits quantizer; therefore we code 8 output states whereas with 12
bits we code 4096 of them.. The diagram showed in figure 1 point out that in the
quantized signal there are 2n -1 = 7 threshold levels. These points are at 0.625-1.875
11
-3.125 -4.375 -5.625 -6.875 -8.125 V. The threshold points have to be set accurately
to divide the range of the signal to quantize in correct quantized signals. The voltages
1.25 -2.5 -3.75 -5 -6.25- 7.5 -8.75 V are the center points of each output code word.
The staircase quantization is the best possible approximation for a straight line
starting from the origin and reaching full scale. The range of the input voltage for
which the same output code is used is called "quantum". In figure 1, the quantum is
1.25 V. In general the quantum is expressed by the relation:
Q = FSR 1 2n = Full scale range 1 2 n
It is plain that the quantum is the lowest analog difference that can be discriminated at
the output. In case of a 12 bits quantizer, still with a full scale of 10 V the quantum is:
Q = 10/212 = 10/4096 = .00244 V = 2.44 mV
In figure 1 is also reported a curve called "quantizing error". This represents the error
that could not be eliminated, due to quantization. Increasing the bit number can
reduce the error. The uncertainty due to quantization and therefore the error due to
quantization is ::!:: Q/2. An AID converter requires a given time interval, even small,
to perform the quantizing and coding operations. This time is called "conversion
time" and depends on several factors, as the converter resolution, the conversion
technique, the speed of the components used in the converter.
During the conversion the signal can vary. Therefore the conversion speed
for a given application depends both upon the required accuracy and upon how much
rapidly varies the signal to convert. The conversion time is often called improperly
"aperture time" ta (Fig.2)
V(t)
∆V
ta
a
If the signal changes of ~ V during the version, we have an amplitude error ~ V which
is ~ V = t ao dV (t)/dt Where dV(t) /dt is the rate of change with time of the input
signal. A signal has not the same rate of variation everywhere, e.g. a sinusoidal signal
has the biggest derivative at the zero passage.
12
The AID converters use very different techniques to perform the analog/digital
conversion. The conversion speed and the resolution are the two guide criteria in
choosing an AID converter.
One of the more simple A/D converters is the counter one, see fig.3. This circuit uses
a digital counter to control the input of a D/ A converter. The clock pulses are applied
to a counter and the D/ A converter output is increased a step at a time of amplitude
corresponding to that of the less significant bit. A comparator compares the D/ A
converter output with the analog input and stops the clock pulses when the output of
the D/ A converter goes over the input signal. The
counter output is therefore the converted digital word. Let us see in detail how is
performed the conversion. The oscillator is always working. The clock is transmitted
to the counter only when the AND input, connected to the comparator is in the logic
state high (" 1 ") or when the signal to convert is larger than the D/ A converter
output.
CLK Reset
& CTR
MSB
LSB
DAC
Vi
Let us apply a reset control. The counter goes to "0000" and the converter gives zero
voltage at the output. The comparator results at high level at the output, the first pulse
that leads the counter to "0001" and the converter output to 0.625 is let go. The
13
converter remains in the starting state. When the new clock pulse passes the counter
goes to "0010" and the DAC output to 1.25 V. The process continues until the counter
receives the ninth pulse, its state then becomes "1001" and the converter output goes
to 5 .625 V and thus is larger than the voltage to converter.
The comparator output changes its state and so the clock pulses are not transmitted to
the counter any more. The word "1001" is then the digital conversion of 5.0 V. The
used converter allows the conversion of positive signals included in the range from 0
to 5 V and has a maximum resolution of 0.625 V. The counter converter is
conceptually simple and its realization is of little difficulty and cheap. On the other
hand it is rather slow in the conversion. Moreover the conversion time is not constant
but depends upon the value to convert.
Successive approximation A/D converter the previous converter is simple but slow.
We can obtain largest conversion :) speeds to the detriment of simplicity. It is the case
of successive; approximation converters of which the board reports an example in an
Integrated version.
The DAC performs the conversion in n steps where n is the converter J settlement in
bits. The working principle of this converter is analogous to that :) of weighing an
object on a laboratory balance, using standard weights as :J reference, according to
the binary sequence 1,1/4,1/8, 1/16 1/n () kilograms. To perform accurately the
measure, we have to start with the largest standard weight and go on in decreasing
order till the one of smallest weight. We place the largest weight; if the balance
doesn't tip we leave the weight and add the one from the others of largest weight. If
the balance doesn't tip we go on. If instead it does tip, we remove the largest weight
added and we replace it with the next one. After having tried n standard weights, the
weighing operation stops. The total of the standard weights remaining on the balance
is the closest approximation to the unknown.
We start by setting the bit 3, thus "1000". The corresponding voltage is worth 5 V and
being lower than the voltage to convert we leave it at "1 ". We set to "1" the bit 2 too.
The word turns into "1100" corresponding to 7.5 V which is larger than the voltage to
convert. We put therefore at zero the bit 2 and we proceed to the bit 1 and set it to "1".
The corresponding word "1010" is worth 6.25 V still larger than 5.0 V, we put than at
zero the bit 1 and set to "1" the bit O. The word turns into "1001" and is worth 5.625
V. This value is still larger than 5.0 V and so the reading jams at "1000" which is the
digital conversion of the voltage 5.0 V. We can see as in this case we have done only
four comparisons to perform the conversion. In the case of an
n bits converter are required n comparisons instead of the 2n foreseen, in the worst
case, 'by the previous converter. We have to make a precise statement: while the
counter had convened 5.0 V into the word "1001 ", the successive approximation one
converts into "1000". This is because whereas the counter converter blocks the clock
14
when the digital word goes over the value to convert and so approaches in excess with
the last significant bit, the successive approximation converter approaches in
deficiency.
APPARATUS USED :
SL. ITEM
NO.
1 Trainer Kit
2 multimeter
5 Connecting Wires
Experiment No: 3
15
Experiment Title: Studies of PCM transmitter and receiver.(To measure the bit rate,
bandwidth requirement and distortion factor of the reconstructed signal in presence of
channel noise).
THEORY: PCM is the only digitally encoded modulation technique commonly used
for digital transmission. The term pulse code modulation as it is not really a type of
modulation but rather a form of digital signals. With PCM, the pulses are of fixed
length and fixed amplitude. PCM is a system where a pulse or lack of a pulse within a
1 or a logic 0 condition. PWM, PPM, and PAM are digital but seldom binary, as does
not represent a single binary digit (bit).Fig shows a simplified block diagram of a
single-channel, simplex (only) PCM system. The band pass filter limits the frequency
of the analog input standard voice-band frequency range of 300 Hz to 3000 Hz.
16
Essentially, there are two basic techniques used to perform the natural sampling and
flat-top sampling. Natural sampling is shown is when tops of the sample pulses retain
their natural shape during the interval, making it difficult for an ADC to convert the
sample to a PCM , the frequency spectrum of the sampled output is different . The
amplitude of the frequency components produced from sample pulses decreases for
the higher harmonics in a (sin x)/x manner Pi information frequency spectrum
requiring the use of frequency equalizer) before recovery by a low-pass filter.
The most common method used for sampling voice signals in top sampling, which is
accomplished in sample-and-hold circuit. The sample and-hold circuit is to sample
the continually changing. Convert those samples to a series of constant-amplitude
PAM volt, the input voltage is sampled with a narrow pulse and then hold until the
next sample is taken. Figure 6-4 shows flat-top sampling, 4 shows, the sampling
process alters the frequency spectrum and introduced aperture error, which is when
the amplitude of the sampled signal pulse time.
This prevents the recovery circuit in the PCM receive) Inducing the original analog
signal voltage. The magnitude of error of the analog signal voltage changes while the
sample is being taken and the width ' of the sample pulse. Flat-top sampling, however,
introduces fewer apertures natural sampling and operates with a slower analog-to-
digital converter.
18
APPARATUS USED:
SL. ITEM
NO.
1 Trainer Kit
2 multimeter
3 Connecting Wires
4 CRO
5 Function Generator
19
20
Experiment No: 4
Objective:
Study of Carrier Modulation Techniques by Phase Shift Keying method.
THEORY
Phase-shift keying (PSK) is a digital modulation scheme that conveys data by
changing, or modulating, the phase of a reference signal (the carrier wave).
Any digital modulation scheme uses a finite number of distinct signals to
represent digital data. In the case of PSK, a finite number of phases are used. Each
of these phases is assigned a unique pattern of binary bits. Usually, each phase
encodes an equal number of bits. Each pattern of bits forms the symbol that is
represented by the particular phase. The demodulator, which is designed
specifically for the symbol-set used by the modulator, determines the phase of the
received signal and maps it back to the symbol it represents, thus recovering the
original data. This requires the receiver to be able to compare the phase of the
received signal to a reference signal — such a system is termed coherent.
In Phase Shift Keying (PSK) modulation techniques, the modulated output
switches between in-phase and out-of phase component of the carrier frequency
for all 'one' to 'zero' transitions. The carrier frequency chosen for PSK modulation
are 1 MHz (o Degree) and 1 MHz (180 Degree)
CARRIER GENERA TOR block on DCL-05 generates the carrier waves 1 MHz
(0 Degree) and 1 MHz (180 Degree), which are available at SIN 2 and SIN 3 post.
The PSK modulator is also built around the 2 to 1 Ana log Multiplexer which
switches between the 1 MHz (0 Degree) and 1 MHz (180 Degree) signals for all
'one' to 'zero" transitions occurring in the transmitted data stream.
The phase detector works in the principle of squaring loops. First step in PSK
detection is the sine to square wave conversion using an Schmitt trigger. This
enables the PSK detector to be built around digital IC's. The Biphase splitter
basically doubles the frequency component of the modulated data and also ensures
that the out of phase component of the modulation signal does not reach the PLL.
The PLL recovers the carrier frequency from the output of the phase splitter, but
the frequency of the recovered carrier is twice that of the transmitted carrier. So a
Divide by 2 counter is used to divide the frequency of the PLL output by 2, thus
recovering the reference carrier. The delay flip-flop is used to compare the phase
of the incoming data and the reference carrier thereby recovering the data.
21
22
Procedure:
1. Refer to the block diagram and carry out the following connections and switch
settings.
2. Connect power supply in proper polarity to the kits DCL-O5 and DCL-O6 and
switch it on.
3. Connect CLOCK and DATA generated on DCL-O5 to CODING CLOCK IN
and DATA INPUT respectively by means of the patch-chords provided.
4. Note: 8th switch of SW1 from data pattern generator is not used to generate
data pattern, output remains at low level irrespective of this switch position.
5. Connect the NRZ-L data input to the CONTROL INPUT of the Carrier
Modulator logic.
6. Connect carrier component SIN 2 to INPUT1 and SIN 3 to INPUT2 of the
Carrier Modulator Logic.
7. Connect PSK modulated signal MODULATOR OUTPUT on DCL-O5 to the
PSK IN of the PSK DEMODULA TOR on DCL-O6.
8. Observe various waveforms as mentioned below.
.Observation:
Observe the following waveforms on oscilloscope and plot it on the paper.
On Kit DCL-O5
1. Input NRZ-L Data at CONTROL INPUT.
2. Carrier frequency SIN 2 and SIN 3.
3. PSK modulated signal at MODULATOR OUTPUT.
On Kit DCL-O6
1. PSK Modulated signal at PSK IN.
2. PSK Demodulated signal at PSK OUT.
3. Observe output of SINE TO SQUEARE CONVERTOR, SQUARING
LOOP, DIVIDE BY 2 on test points provided.
23
Waveforms:
24
Apparatus Required:
• Experimental Kits Data Conditioning and Carrier Modulation (DCL-05) &
Data Reconditioning and Carrier Demodulation (DCL-06).
• Power Supply
• 20 MHZ Dual Trace Oscilloscope
• Multimeter
• Patch Chords
25
Experiment No: 5
Theory:
In Frequency Shift Keying modulation techniques, the modulated output shifts
between two frequencies for all 'one' (mark) to 'zero' (space) transitions. The
carrier frequency chosen for FSK modulation is 500 KHz and 1 MHz. Note that
the above frequencies are greater than twice the modulating frequency. Note that
the FSK may be thought of as an FM system in which the carrier frequency is
midway between the mark and space frequencies, and modulation is by a square
wave.
CARRIER GENERATOR block on DCL-O5 generates the carrier waves 500KHz
and1 MHz, which are available at SIN1 and SIN2 post.
The FSK modulator is also built around the 2 to 1 Analog Multiplexer which
switches between the 500KHz and 1 MHz signals for all 'one' to 'zero" transitions.
FSK describes the modulation of a carrier (or two carriers) by using a different
frequency for a 1 or 0. The resultant modulated signal may be regarded as the sum
of two amplitude modulated signals of different carrier frequency
Procedure:
1. Refer to the block diagram and carry out the following connections and switch
settings.
2. Connect power supply in proper polarity to the kits DCL-O5 and DCL-O6 and
switch it on
3. Connect CLOCK and DATA generated on DCL-O5 to CODING CLOCK IN
and DATA INPUT respectively by means of the patch-chords provided.
4. Note: 8th switch of SW1 from data pattern generator is not used to generate
data pattern, output remains at low level irrespective of this switch position.
5. Connect the NRZ-L data input to the CONTROL INPUT of the Carrier
Modulator logic.
6. Connect carrier component SIN 1 to INPUT1 and SIN 2 to INPuT2 of the
Carrier Modulator Logic.
26
Observation:
Observe the following waveforms on oscilloscope and plot it on the paper.
On Kit DCL-O5
1. Input NRZ-L Data at CONTROL INPUT.
2. Carrier frequency SIN 1 and SIN 2.
3. FSK modulated signal at MODULATOR OUTPUT.
On Kit DCL-O6
1. FSK Modulated signal at FSK IN.
2. FSK Demodulated signal at FSK OUT.
3. Observe output of PHASE DETECTOR, LPF, VCO on test points provided.
Experimental Data:
NOTE: In FSK demodulator PLL circuit used is very sensitive to input voltage
level, because of which you may get blurred output signal if input power varies
slightly. To get clear signal at the output tune pot P3 in FSK Demodulator section.
To get better results set the following bit pattern for INPUT DATA:
10101010
10101110
11101010
00111010
11001100
27
28
Waveforms:
Analysis:
A small phase lag exist between the modulating data and the recovered data
because of the limitation of tracking ability and the time response of PLL
29
Switch Faults:
Note: Keep the connections as per the procedure. Now switch corresponding
fault switch button in ON condition & observe the different effect on the output.
The faults are normally used one at a time.
1. Put switch 1 of SF1 (DCL-O5) in Switch Fault section to ON position. This
will open capacitor for filtering of SIN 1 signal. Sine wave SIN 1 will be
distorted and its amplitude gets reduced.
2. Put switch 2 of SF1 (DCL-O5) in Switch Fault section to ON position. This
VI/ill disable channel selection signal going to Modulator IC. Modulator
output contains only single channel (INPUT 1) data.
3. Put switch 2 of SF1 (DCL-O6) in Switch Fault section to ON position. This
will remove resistor connected to PLL in FSK demodulator. Center frequency
of PLL changes and output of FSK demodulator gets distorted.
30
EXPERIMENT NO: 6
Objective:
Study of Carrier Modulation Techniques by Amplitude Shift Keying method.
Theory:
Carrier modulation is a technique by which digital data is made to modulate a
continuous wave (sine wave) carrier. For all types of carrier modulation, the
carrier frequency should be at least 2 times that of modulating frequency. In
Amplitude shift keying, the carrier is transmitted when the modulating data is 'one'
and the carrier is rejected from transmission when the data is 'zero'. In DCL-O5
the ASK Modulators employs an Analog Multiplexer as a modulating switch,
which can switch between carrier and ground, for every 'one' to 'zero' transitions.
The carrier frequency chosen for ASK modulation is 1 MHz.
ASK DEMODULA TOR block on DCL-O6 employs an envelope detector to
recover the data from the modulated carrier. The ASK modulated input is fed to
the half wave rectifier. The rectified input is fed to the filter, where the original
data is recovered. The threshold detector is used to recover the original amplitude
levels of the data. So whenever the sinewave is transmitted, the detector identifies
it as a 'one' and whenever the carrier is absent, the detector identifies it as a 'zero'.
ASK describes the technique the carrier wave is multiplied by the digital signal f
(t). Mathematically, the modulated carrier signal s (t) is:
s (t) = f(t)sin(2π fC t + Φ)
31
Apparatus Required:
• Experimental Kits Data Conditioning and Carrier Modulation (DCL-05) &
Data Reconditioning and Carrier Demodulation (DCL-06).
• Power Supply
• 20 MHZ Dual Trace Oscilloscope
• Multimeter
• Patch Chords
Observation:
Observe the following waveforms on oscilloscope and plot it on the paper.
On Kit DCL- O5
1. Input NRZ-L Data at CONTROL INPUT.
2. Carrier frequency SIN 2.
3. ASK modulated signal at MODULATOR OUTPUT.
On Kit DCL-O6
1. ASK Modulated signal at ASK IN.
32
Waveforms:
Analysis:
It has been observed that a very small time lag between the modulating data and
the recovered data
Switch Faults:
33
Note: Keep the connections as per the procedure. Now switch corresponding
fault switch button in ON condition & observe the different effect on the output.
The faults are normally used one at a time.
1. Put switch 2 of SF1 (DCL-O5) in Switch Fault section to ON position. This
will disable channel selection signal going to Modulator IC. Modulator output
contains only single channel (INPUT 1) data.
34
Experiment No. – 7
Experiment Title: Studies on Delta modulator & Demodulator using trainer kits.
OBJECTIVE: This experiment will enable one to understand how a Delta Modulator
Circuit can work to have a good result w.r.t PCM.This experiment will also help to
understand the operation of Digital modulation technique.
THEORY
Delta Modulation is a system of digital modulation developed after pulse code
modulation. In this system, at each sampling time, say the Kth sampling time, the
difference between the sample value at sampling time K and the sample value at the
previous sampling time (K-1) is encoded into just a single bit. i.e. at each sampling
time we ask simple question:
Has the signal amplitude increased or decreased since the last sample was taken?
If signal amplitude has increased, then modulators output is at logic level 1.
If the signal amplitude has decreased, the modulator output is at logic level 0.
Thus the output from the modulator is a series of zeros and ones to indicate rise and
fall of the waveform since the previous value.
One way in which Delta modulator and demodulator is assembled is as show in Fig.
35
36
Suppose at some time-instance t = 0, the integrator output voltage is lower than the
analog input. This causes the voltage comparator voltage to go high i.e. logic '1'. This
data is latched in the D-Flip-Flop at the rising edge of Transmitter clock. The latched
'1'output of D-flip is translated to - 4V by the Unipolar to Bipolar converter block.
The integrator then ramps up to catch analog signal.
At the next clock cycle t = 1, the integrator output becomes more than the analog
input, so a '0' is latched into D-Flip-Flop. The integrator now ramps downward as +
4V voltage signal from Unipolar to Bipolar converter appears at its input. Thus the
ramp signal again tries to catch the fallen analog signal.
37
As we can observe, after several clock cycles the integrator output is approximation of
the analog input which tries to catch up the analog input at each sample time. The data
stream from D-flip-flop is the Delta Modulators Output.
The Delta demodulator consists of a D-Flip-Flop a Unipolar to Bipolar converter
followed by an integrator and a Low Pass Filter. The Delta demodulator receives the
data from D-Flip-Flop of Delta Modulator. It latches this data at every rising edge of
receiver clock which is delayed by half clock period with respect to transmitter clock.
This has been done so that the data from transmitter may settle down before being
latched into the receiver Flip-Flop.
The Unipolar to Bipolar converter changes the output from D-FIip-Flop to either - 4V
or + 4V for logic T and '0' respectively.
As it has been seen in case of modulator when the output from unipolar to bipolar
converter is applied to integrator, its output tries to follow the analog signal in ramp
fashion and hence is a good approximation of the signal itself. The integrator's output
contains sharp edges which are 'smoothened out' by the Low - Pass Filter, whose cut-
off frequency is just above the audio band.
38
Distortion in delta modulation occurs due to, following causes: 1) Information Loss
due to rapid input changes.2) Overloading
As it has been seen, when the analog signal is greater then the integrator output,
the integrator ramps up to meet the analog signal. The ramping rate of integrator is
constant. Therefore if the rate of change of analog input is faster than the ramping
rate, the modulator is unable to catch up with the information signal. This causes a
large disparity between the information signal and its quantized approximation. This
error / phenomenon is known as slope - overloading and causes the loss of rapidly
changing the information.
At first it may look as though the problem of slope overloading can be solved
increasing the ramping rate of the integrator. But as it can be seen from the figure the
effect of the large step-size is to add large sharp edges at the integrator output and
hence it adds to noise problem faced at receiver. This effect itself leads to distorted
receiver output. Increasing sampling rate cannot be the solution to the slope -
overloading problem as it determines how fast the samples are taken and not the
ramping rate of the integrator.
39
(C) Another problem with delta modulation is that it unable to pass DC information.
This is not a serious limitation of speech communication but for the systems like
video (picture) Transmission DC level does not provide information about brightness
level of the picture for more details see page no. 40.
The above stated limitations of the delta modulation may be traded for acceptable
price in speech application but is totally unsatisfactory for music or video signals.
APPARATUS USED:
SL. ITEM
NO.
1 Trainer Kit
2 CRO
5 Connecting Wire
40
Experiment No. –8
OBJECTIVE: This experiment will enable one to understand how an Adaptive Delta
Modulator Circuit can work to have a good result w.r.t delta modulator. This
experiment will also help to understand the operation of Digital modulation technique.
THEORY
As it has been seen, delta modulation system is unable to chase the rapidly changing
information of the analog signal which gives rise to distortion & hence poor quality
reception. This is known as slope overloading phenomenon. The problem can be
overcomed by increasing the integrator gain (i.e. step-size). But using high step-size
integrator would lead to a high quantization noise.
QUANTISATION NOISE:
It is defined as error introduced between the original signal & the quantized signal due
to the fixed step size in which the signal (quantized) is incremented. As the error is
random in nature & hence unpredictable, it can be treated as noise. High quantization
noise may play havoc on small amplitude signals. The solution to this problem is to
increase the integrator gain for fast-changing input & to use normal gain for small
amplitude signals.
The basic idea is to increase the integrator the integrator gain (it is doubled on this
trainer) when slope overload occurs. If still it is unable to catch up with the signal, the
integrator gain is doubled again. The integrator on board has four available gains
standard, standard X2, standard X4, and standard X8. The integrator thus adopts itself
to the gain where its lowest value can just overcome the slope overloading effect. See
fig. 6A.
the control circuit its output goes low. The counter now advance to 01 doubling the
integrator gain. This increases the ramping rate of the integrator & it is able to catch
the analog signal faster. In the next clock cycle if the same situation continues the
counter advances to '10' thus forcing the integrator gain to quadruple its standard
value. This situation continues till the counter advances to ' 11' where it remains
locked until the control logic does not detect a change in the bit level at its input
As soon as the control circuit detects a change in the bit level, its output goes high,
thus resetting the counter & thus normalizing the integrator gain.
In Adaptive Delta Demodulator the control circuit receives the same bit stream as the
transmitted one except for the fact that it is received after a half clock cycle delay.
The functioning of the Receiver's control circuit & counter is same as that of the
transmitter's block. Therefore, the demodulator’s output which itself is a good
approximation of the analog input signal accepts for the inherent spikes. The output
from integrator is passed to a Low Pass Filter to 'smooth out 'the waveform. Thus,
Adaptive Delta Modulation system is thus able to reduce slope-over load error at an
expense of small increase in quantization error. It turns out that in matter of speech
transmission the reduced slope error provides a net advantage in spite of slight
increase in quantization error & that the adaptive delta.
41
FIG A
42
FIG B
43
namely, the control circuit & the counter. They function in the same way as in
modulator part, except for the fact that they are clocked by the receiver clock.
Consider the Adaptive Delta Modulator in operation. In normal case, when slope
overloading is not occurring, the integrator output always hunt above & below the
analog input even after it has caught up with it. The output from the D-Flip-Flop is a
constantly changing ' 1' to '0' at each TX.CLOCK edge. Even when the analog input is
changing at a slightly higher rate, the integrator ramp output is able to catch it in two
clock cycles. Thus the output of the D-Flip-Flop is never a three or more consecutive
'0' or T s.
The changing input to the control circuit ensures that its output to the counter is high
& hence the counter is 'rested' at every clock cycle. Thus the control word from
counter is always '00' forcing the integrator gain at its lowest value, thereby reducing
quantization noise. Here the Adaptive Delta modulator is behaving just as a Delta
Modulator. Suppose, now a fast changing analog signal appears at the input of the
modulator such that the slope overloading occur. The integrator output no longer
follows the analog signal but it spends its time trying to catch up the analog signal
(either it ramps down or up continuously). As a result of continuous ramping in one
direction, the D-Flip-Flops output is either '0' or * 1' for three or more consecutive
time. As soon as the third continuous 1 / 0 is sensed by
APPARATUS USED:
SL. ITEM
NO.
1 Trainer Kit
2 CRO
5 Connecting Wire
PROCEDURES:
44
Experiment No. –9
OBJECTIVE: This experiment will enable one to understand how an Delta Sigma
Modulator Circuit can work to have a good result(to pass DC level signal which is a
must in video system w.r.t delta modulator).This experiment will also help to
understand the operation of Digital modulation technique.
THEORY
The Delta Modulation & Adaptive Delta Modulation suffers from two serious
limitations, namely,
(1) They are not able to pass DC. level information which is a must in video systems.
(2) The signal - to - noise ratio decreases as the signal frequency increases.
Both these serious limitations can be overcomed by Delta Sigma modulation.
The Delta Modulator on-board can pass DC level information if the integrators are
zeroed before hand (by level adjust presets) & the gains of integrator at two ends (TX
& RX) are exactly equal. This is possible on ST2105 board but not in real life
operation where TX & RX. are separated by hundreds of kms. & they are expected to
work over a long period without additional setting up.
The inherent short coming of the Delta Modulator to pass DC level information is due
to the fact that it passes the information about the change in the voltage level & not
the actual voltage level itself. It encodes the change from -2V to -IV same as it would
encode a change from +2V to +3V. J
In Delta Sigma Modulation the integrator is added hi front of the Delta sigma
modulator. This simple arrangement makes a big change in the circuit behaviors. It
now responds to actual voltage levels rather than the change in the voltage level.
The effect of the addition of integrator before the delta modulator can be done away
by adding a differentiator at the end of the receiver. The System looks as shown in
45
The system shown in the Figures can be simplified further. Since the voltage
comparator is highly non-linear device cannot transfer the two integrations at its input
to one integration at its output.
The voltage comparator functionally can be thought of as a unity gain differential
amplifier followed by a zero-crossing detector.
If the voltage comparator is replaced by this circuit, the no. of integrators can be
reduced to one (at the output of unit-gain differential amplifier).Similarly, since the
low pass filter is a linear circuit element the effect of integrator at the input of
L.P.f...is nullified by the differentiator present at its output. Therefore these two
blocks are redundant in the receiver.
The resulting circuit looks as in Fig. 9A & 9B.
46
To see how the circuit functions, suppose that initially the TX.DATA is at logic * 1'
& that input remains constant at+2V.
Since logic 1 is present at the input of Unipolar-to-Bipolar converter, its output is-4V,
which appear at-VE terminal of unity gain differential amplifier. Its output soon
switches to+6V. This +6V appears at the input of integrator which ramps down at a
fast rate. As soon as it crosses zero level, the zero-crossing detectors output goes zero
47
which is latched in the coming clock cycle. Thus the D-Flip-Flops output goes low &
a zero is transmitted.
The zero's appearance at input of unipolar to bipolar converter makes it switch its
output to +4V. With +4V input to the unity gain differential amplifier, its output goes
to -2V level. Since -2V signal is present at input of integrator; it ramps up slowly, as
the rate of ramping depends on the input applied to it. The ramp will not be able to
cross-up zero at this slow rate, so again a zero is latched at the next clock cycle. Thus
the inference which can be drawn is that smaller the signal at integrator input, slower
it will ramp up or down thus latching the same data bits for more clock periods.
Thus the mark to space ratio conveys the information about the mean level of the
signal & the signal & the modulator's output depends on the actual signal amplitude
itself & not only on its change.
The received data is latched into D-Flip-Flop by the Receiver clock which has been
before being latched. The level changer provides the +VE & -VE voltages needed to
reconstruct the input analog signal. It also inverts the signal & compensate for the
inversion at transmitter automatically. The work of the low pass filter is to remove the
spikes introduced by switching action of unipolar to bipolar converter.
The received data is latched into D-Flip-Flop by the Receiver clock which has been
before being latched. The level changer provides the +VE & -VE voltages needed to
reconstruct the input analog signal. It also inverts the signal & compensate for the
inversion at transmitter automatically. The work of the low pass filter is to remove the
spikes introduced by switching action of unipolar to bipolar converter.
APPARATUS USED:
SL. ITEM
NO.
1 Trainer Kit
2 CRO
5 Connecting Wires
48
49
50