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Reconstruction Trainer
ST2101
Operating Manual
Ver 1.1
ST2101
ST2101
1.
Features
2.
Technical Specifications
3.
Theory
4.
5.
Nyquist Criterion
10
6.
11
7.
12
8.
14
9.
17
Experiment 1
Study of Signal Sampling and Reconstruction Techniques
Study of Nyquist Criteria and Aliasing
26
Experiment 2
i) Study of the effect of Sample /Hold Circuitry on Reconstructed
Waveform
29
i)
ii)
ii)
Experiment 3
Comparison of Frequency Response of 2nd order and 4th order
Butterworth Low Pass Filter
32
i)
ii)
10.
Warranty
34
11.
List of Accessories
34
ST2101
Features
RoHS Compliance
ST2101
Technical Specifications
Crystal Frequency
6.4 MHz
Sampling Frequency
On-board Generator
Duty cycle
Test Point
51 in numbers
Interconnections
4 mm sockets
Power Consumptions
3 VA (approximately)
Power Supply
Dimensions (mm)
Weight
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Theory
The aim of any communication system is to transmit information from one location to
another. In case of voice communication, this information will be speech.
The signal which contains the information to be transmitted is known as information
signal and in the case of voice communication this will be a continuously changing
signal containing speech information. The aim is to reproduce this information signal
as accurately as possible, at the distant, receiving end of the communication system.
In the exercises to follow, you will simulate audio signal by a 1 KHz test signal
provided On-board. The repetitive, non-changing (in amplitude, frequency or phase),
waveform does not contain information, but that does not mean we cannot use it.
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 communications.
Theory of Sampling :
In analog communication systems like AM, FM, the instantaneous value of the
information signal is used to hang certain parameter of the carrier signal.
Pulse modulation systems differ from these systems in a way that they transmit a
limited number. of discrete states of a signal at a predetermined time; sampling can be
defined as measuring the value of an information signal at predetermined time
intervals. The rate at which the signal is sampled is known as the sampling rate or
sampling frequency. It is the major parameter which decides the quality of the
reproduced signal. If the signal is sampled quite frequently (whose limit is specified
by Nyquist Criterion), then it can be reproduced exactly at the receiver with no
distortion.
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Sampled Output
Figure 2
PAM is an analog system because the amplitude of pulse can vary infinitely i.e.the
levels is not discrete.
An information signal sent through an ideal switch which is operated by a control
signal, isolated from the information signal, produces a PAM signal. When the switch
is open, the voltage is zero; when switch is closed the output voltage is equal to the
instantaneous signal voltage. The sample width depends upon how long a switch
remains closed.
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Figure 3
As it can be seen from above figure 3 the multiplier output has some value as m (t)
when the pulse occurs, otherwise it is zero. Rectangular waveforms can represented as
summation of Sine/Cosine waveforms of fundamental frequency plus infinite number
of harmonics.
Since PAM is amplitude modulation of pulse, we expect the sidebands to be formed
around fundamental frequency and each harmonics.
If the sampling frequency is Fs, the frequency spectrum of the PAM signal is as
shown below figure 4.
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The PAM signal contains the spectrum of the base band signal unlike that in
and where it is absent it is due to this fact that we can recover the original
signal.
2.
In AM, a fixed amplitude carrier component is also present at the unmodulated frequency fc. In PAM no such component exists in PAM spectrum.
The information signal can be recovered from the PAM signal by using a low pass
filter of cut-off frequency FM.
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Nyquist Criterion
As shown-in the figure 5 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.
Figure 5
Nyquist Criterion (Sampling Theorem)
The Nyquist criterion 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.
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The narrower pulses allow us to time division multiplex many such PAM
signals i.e. we can send many no. of PAM signals over same channel at a time.
Hence lower duty cycle beneficial in this respect.
2.
The narrower pulses has wider frequency spectrum. Hence the wider bandwidth
channel is required.
3.
Narrower pulses have less power as the power content of a pulse depends on its
amplitude and width. During transmission and demodulation the inherent noise
can play a major havoc on the low power signal. Hence a pulse of larger dutycycle is desirous for this sake.
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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 reduce to zero because on application of finite time taken by a switch to closet
open on application of the hold signal. Therefore a small value of aperture time is
sought after.
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.
Drop 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 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.
Scientech Technologies Pvt. Ltd.
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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.
Aliasing :
If the signal is sampled at a rate lower than stated by Nyquist criterion, then there is
an overlap between the information signal and the sidebands of the harmonics. Thus
the higher and the lower frequency components get mixed and cause unwanted signals
to appear at the demodulator output. This phenomenon is turned as aliasing or fold
over distortion. The various reasons for aiasing and its prevention are as described.
1.
If the signal is sampled at rate lower than 2Fm then it causes aliasing. Let us assume a
sinusoidal waveform of frequency FIN which is being sampled at rate Fs < 2Fm. In the
figure 9 dots represents the sample points.
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2.
The system is designed to take samples at frequency slightly greater than that stated
by Nyquist rate. If higher frequencies are ever present in the information signal or it is
affected by high frequency noise then the aliasing will occur.
This does not generally happen in properly designed telephone network where speech
channels are band-limited by filters before sampling.
In control engineering and telemetry, however, out of band high frequencies either
from source or due to noise pick-up can be present. In this case band-limiting filters,
generally known as anti-aliasing filters are usually installed prior to sampling to
prevent aliasing.
As a principle, the system is designed to sample at rate higher than the rate to take
into account the equipment tolerances, ageing and filter response.
3.
Roll-off is a term applied to the cut-off gradient of a filter. No filter is ideal and
therefore frequencies above the nominal cut-off frequency may still have significant
amplitudes at a filter's output. If proper sampling rate and appropriate filter response
is not chosen, aliasing will occur.
4.
If very small duty cycle is used in sample-and-hold circuit aliasing may occur if the
signal has been affected by noise. High frequency noises generally mix with the
high frequency component of the signal and hence causes undesirable frequency
components to be present at the output.
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The ratio of VOUT: VIN is known as Transfer function for the circuit. For RC low pass
filter, the transfer function can be derived by using potential divider resistance.
So,
If
V OUT
1
-------- =
---------------VIN
1+2R2C2
1
V OUT 1
= ------ then
--------- = ----= 0.707 = -3db
RC
V IN
2
This is the half-power point of the filter i.e. at frequency = RC, the output power
decreases to half of the input power. This is also known as the cut-off frequency (Fc).
The filter not only causes amplitude change but a change in phase is also experienced.
A typical response of a low pass filter is as shown below:
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2.
3.
The active filters employ transistors or op-amps in addition to resistor and capacitor.
The resistors at the output of the op-amp create a non-inverting voltage amplifier of
voltage gain K while other resistor and capacitor sets the frequency response
properties of the filter.
An ideal filter should have zero loss in pass band and infinite loss in stop band. In
practice no ideal response exists, but there are many responses which approximate the
ideal response namely,
Butterworth, Chebyshev, Bessel etc. the comparison of these filter responses are as
shown in figure 13.
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fall-off Octave
First
Second
Fourth
See figure 15.
6
12
14
fall-off decade
20
40
80
Phase at cut-off
frequency
- 45
- 90
- 180
1
---------------1+ (f / fc)2n
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Figure 15
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Frequency (Normalized)
Amplitude Vs Frequency & Phase Vs Frequency
Response of Second Order Butterworth Low Pass Filter
Figure 16
For this circuit, the voltage gain has been set equal to 1.586.
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Though increasing order filter is desirable, there is a price that we have to pay for
steeper fall-off.
1.
2.
Increase in order increases phase lag, though it is not so critical in audio circuits.
2.
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Experiment 1
Objective :
1. Study of signal sampling and reconstruction techniques
2. Study of Nyquist criteria and aliasing.
Procedure :
1.
Connect the mains plug into the mains board. Keep the power switch in Off
position.
2.
3.
4.
5.
6.
7.
Display 1 KHz sine wave (TP12) and Sample Output (TP37) on an oscilloscope.
The display shows 1KHz. Sine wave being sampled at 32 KHz, so there are 32
samples for every cycle of the sine wave. (figure 19)
Link the Sample output to Fourth Order low pass Filter display Sample Output
(TP37) and the output of filter (TP46) on the oscilloscope. The display shows
the reconstructed original 1 KHz sine wave. (figure 20)
8.
9.
10.
So far, we have used sampling frequencies greater than twice the maximum
input frequency. To study Nyquist criteria, set sampling rate of 8KHz, 50% duty
cycle.
11.
Remove the link from 1 KHz sine wave output to the Signal Input.
12.
Obtain a 2V peak, 2KHz Sine wave from 600 ohms output of the function
generator to Signal Input. Observe the waveform at Signal Input and fourth
order low pass filter output (TP46). Observe the two are similar but the second
logging in phase. This is as expected from filters phase/ frequency response.
13.
Decrease the sampling rate to 32 KHz and then to 2KHz. Observe the distorted
waveform at filter's output (TP46). This is due to the fact that we under-sampled
the input waveform overlooking the Nyquist criteria and thus the output was
distorted even though the signal lie below the cut-off frequency of the filter.
This also describes the phenomenon of Aliasing.
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Figure 19
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Figure 20
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Experiment 2
Objective :
1. Study of the effect of Sample /Hold circuitry on reconstructed waveform
2. Effect of sampling pulse duty cycle on the reconstructed waveform in
sample and sample hold output
Procedure :
1.
2.
Turning 'On' the trainer, selects 32 KHz sampling rate by default. Observe the
Sample output (TP37) and the Fourth Order low pass filters output (TP46).
3.
Vary the position of Duty Cycle Selector switch from 0% to 90% (position 0 to
9), Observing how the Sample Output changes and how the amplitude of filter's
output changes. This amplitude increase; in early as the duty cycle is increased
from 10% to 90%.
4.
Disconnect the Sample Output from filter input. Link Sample & Hold output to
Fourth Order low pass Filter's Input. Set the Duty Cycle Selector switch to
position '5' (50%). (figure 21)
5.
Observe the waveform at Sample & Hold output (TP39) on oscilloscope. Vary
the sampling frequency to illustrate how each sample is held at the sample/hold
output. Also observe the filter output at TP46. (figure 22)
6.
Vary the sampling pulse duty cycle from 0% to 90% and note that in contrast to
step 3, the filter's output amplitude is now independent of the sampling duty
cycle and is equal to the amplitude of the original signal input. This is an
important result - with Sample and Hold Output, the proportion of sampling
time to holding time has no effect on reconstructed waveform provided that
Nyquist criteria has been followed. In practical digital communication, this
result is very useful as the use of narrow pulses let many channels to be
multiplexed with maximum amplitude of reconstructed signal if sample/hold
feature is utilized in communication system.
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Figure 21
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Figure 22
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Experiment 3
Objective :
1. Comparison of frequency response of 2nd order and 4th order.
2. Butterworth low pass filter.
Procedure :
1.
2.
Link 1 KHz Sine wave output to Signal Input. Add a link from SAMPLE output
to input of the Second Order Low Pass Filter and to the input of Fourth Order
Low Pass Filter. Observe the outputs of two filters (TP42 and 46 respectively)
on the oscilloscope. Vary the sampling frequency with duty cycle set at 50%.
Compare the output of filter in each case. Note, that the output of fourth order
filter always exhibits less distortion than second order filter. This is because
fourth order filter has a sharper roll-off and thus rejects (attenuates) more
unwanted frequency components caused by sampling. (figure 23).
Also compare the phase lag between input and output for each filter.
3.
4.
Repeat the above procedure with sample and hold circuitry. Does the output
exhibits less distortion as compared to output in case of sample circuitry?
5.
6.
7.
Repeat the above steps with fourth order filter, which one of the two filter's cutoff has more gradient? What is the phase lag input and output?
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Figure 23
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Warranty
1.
We guarantee the 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 product is not operated as per the instruction given in the operating
manual.
b)
The agreed payment terms and other conditions of sale are not followed.
c)
d)
3.
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 Accessories
1.
2.
3.
4.
5.
6.
34