Experiment 12 – Sampling and Aliasing

Achievements in this experiment

You will be able to intuitively visualize the spectrum of a sampled signal, and aliasing. You will be
able to use this to gain an intuitive understanding of sampling theorems for minimum sampling
rates.

Preliminary discussion

The conversion of analog signal to digital format involves two stages:

- first, the capture of "frozen" sample values
- then, the digitization of these frozen analog values

Further processing may be applied to improve the storage efficiency, ie, to reduce the memory
needed to a minimum.

In this lab we are concerned only with the sampling process. It is evident that the choice of
sampling rate is the paramount issue: Too slow means that some details are lost with samples
too far apart. If the sample spacing is too fine, resources are wasted, i.e. storage and
processing time. A suitable balance between these considerations is needed.

You will start with the sampling of some typical signals, then observe the recovery of the
continuous-time signals from sample sequences at various rates. From this you will be able to
discover the link between the minimum sampling rate and bandwidth.

This lab opens the door to gaining an intuitive understanding of the theory and practical issues
underlying sampling.

In Part 1 you set up sampling operations of selected test signals and carry out observations in
the time domain. Next, you investigate the reverse process, recovering the analog signal, and
examine the effect of various sampling rates.

In Part 2 you retrace the time domain investigations of Part 1 with observations in the
frequency domain. This provides a systematic structure for the processes involved and makes
possible intuitive mathematical interpretation. Equipped with this insight, you will be able to
easily formulate criteria for choosing efficient sampling rates.

12-2 © 2011 Emona Instruments Experiment 12 – Sampling and Aliasing V1.2

Pre-requisite work

Question 1
Look up or derive the trigonometric identity for the product of two sines expressed as a sum.
Confirm that the frequencies in this sum are (f1 + f2) and |f1 - f2|, where f1 and f2 are the
input frequencies. Confirm that the output components are of equal magnitudes.

Question 2
Look up or derive the Fourier series of a squarewave of duty ratio other than 50% (25% and 1%
say). Note the sinx/x shaped spectrum envelope. Locate the frequency of the first null of the
envelope for each case and note the relationship with the pulse width.

Now consider the 50% duty ratio case. Comment on the disappearance of the even harmonics.

Question 3
Derive the spectrum of the product of a sinewave and a 1% duty ratio squarewave. You can do
this easily by using superposition with the results in Question 1 and Question 2. For convenience,
make the frequency of the squarewave around five times the sinewave frequency. Plot the
resulting spectrum.

Experiment 12 – Sampling and Aliasing © 2011 Emona Instruments 12-3

Equipment

PC with LabVIEW Runtime Engine software appropriate for the version being used.

NI ELVIS 2 or 2+ and USB cable to suit

EMONA SIGEx Signal & Systems add-on board

Assorted patch leads

Two BNC – 2mm leads

Procedure
Part A – Setting up the NI ELVIS/SIGEx bundle

1. Turn off the NI ELVIS unit and its Prototyping Board switch.

2. Plug the SIGEx board into the NI ELVIS unit.

Note: This may already have been done for you.

3. Connect the NI ELVIS to the PC using the USB cable.

4. Turn on the PC (if not on already) and wait for it to fully boot up (so that it’s ready to
connect to external USB devices).

5. Turn on the NI ELVIS unit but not the Prototyping Board switch yet. You should observe
the USB light turn on (top right corner of ELVIS unit).The PC may make a sound to indicate that
the ELVIS unit has been detected if the speakers are activated.

6. Turn on the NI ELVIS Prototyping Board switch to power the SIGEx board. Check that
all three power LEDs are on. If not call the instructor for assistance.

7. Launch the SIGEx Main VI.

8. When you’re asked to select a device number, enter the number that corresponds with
the NI ELVIS that you’re using.

9. You’re now ready to work with the NI ELVIS/SIGEx bundle.

10. Select the EXPT 12 tab on the SIGEx SFP.

Note: To stop the SIGEx VI when you’ve finished the experiment, it’s preferable to use the
STOP button on the SIGEx SFP itself rather than the LabVIEW window STOP button at the
top of the window. This will allow the program to conduct an orderly shutdown and close the
various DAQmx channels it has opened.

Ask the instructor to check

your work before continuing.

12-4 © 2011 Emona Instruments Experiment 12 – Sampling and Aliasing V1.2

Experiment

Part 1: through the time domain

11. In this exercise we observe the sampling of a sinewave. Patch up the model in Fig 2.

Fig 1: block diagram for sampling with narrow pulses

Fig 2: SIGEx model for sampling with narrow pulses

The sampling signal is obtained from the FUNCTION GENERATOR square wave output which is
set up specifically to drive the MULTIPLIER block like a switch. When the sampling signal is
non-zero, ie at 1V, the input sinewave is passed. However when the sampling signal is zero volts,
then the input signal is not passed, and 0V is output. This emulates an open/close switch.

Settings are as follows:

FUNCTION GENERATOR: Squarewave selected; 10kHz; 1Vpp, 0.50V Offset, with DUTY
CYCLE=50%
SCOPE: Timebase 2ms; Rising edge trigger on CH0; Trigger level=0V
ANALOG OUTPUT (DAC-1): 4Vpp sinewave at 1kHz

12. View the sinewave input to the MULTIPLIER on CH0, and the sampling signal at the
MULTPLIER on CH1. Confirm that they are as expected.

13. Display the analog input and the output sampled at 10,000 samples/sec. Try one or two
other sampling rates and various DUTY CYCLES settings. These parameters are controlled from

Experiment 12 – Sampling and Aliasing © 2011 Emona Instruments 12-5

the FUNCTION GENERATOR instrument panel. Slippage between signals will occur due to lack
of synchronisation between signals.

This demonstration of the sampling process is evocative because it makes it possible to view the
samples as we imagine them, as individual narrow pulses with amplitude proportional to the
sample value, over the entire width of the pulse. However in most applications further
processing will be involved, such as encoding this sampled level as a digital representation (i.e.,
analog-digital conversion). Hence the sampled value must be held while this is carried out. A
sample-and-hold device is provided in the DISCRETE-TIME section of the SIGEx board.

14. Repeat step 13 using sample and hold. Refer to Fig 4 for the SIGEx model. Compare the
outcome with step 13 .

Fig 3: block diagram for sampling with full width pulses (using Sample/Hold)

Fig 4: SIGEx model for sampling with full width pulses (using S/Hold)

Fig 5: example signals from SIGEx model of Fig 2 & 4

12-6 © 2011 Emona Instruments Experiment 12 – Sampling and Aliasing V1.2

Next we consider the recovery of the original analog waveform from the sample train. We will
use a lowpass filter to smooth out the jagged corners of the stepped signal generated with the
Sample/Hold. This has a good chance of succeeding when variations between samples are
relatively small.

Fig 6: block diagram for recovering the analog signal

15. Attempt the recovery of the analog signal from the stepped sample train from the
SAMPLE/HOLD by means of the TUNEABLE LOWPASS FILTER (Fig 7). Set the Fc tuning knob
to full clockwise. Set the GAIN to give a gain of 1 (knob at mid-range). Start with a high
sampling rate, 10k samples/sec, say. Display the filter output and observe the effect of
reducing the filter bandwidth. Compare this output with the original unsampled signal.

16. Since the sample rate is set by the clock signal, we will interchangeably refer to the
sampling rate as a clock rate, and use the unit “Hz” rather than “samples/sec”.

Question 4
Repeat this for a few other sampling rates, from 10000Hz, down to 2000Hz, say. Document your
readings in Table 1 below. From these observations, what is the minimum sampling rate you
consider adequate to allow recovery of the analog signal without too much distortion, on the
basis of this sampling format (i.e. using the SAMPLE/HOLD function).

Table 1: sample rate readings for recovery from S/H

Sample rate (Hz) TLPF setting Recovered

(approx.position) amplitude (V)

Experiment 12 – Sampling and Aliasing © 2011 Emona Instruments 12-7

Question 5
Repeat the procedures in step 15 for recovery using the TUNEABLE LPF using the sample train
generated with the system in Fig 2, i.e. with narrow pulses. Document your readings in Table 1
below. Compare the outcome with those obtained with the S/Hold method. Do you expect one
of these sample formats to be better for interpolation to analog form? Is this borne out by
your results?

Table 2: sample rate readings for sampled pulse train recovery

Sample rate (Hz) TLPF setting Recovered

(approx.position) amplitude (V)

17. Leave the TLPF Fc control set as per the last few results, for the next few questions.

Question 6
Examine the step and impulse responses of the filter at the settings that give you the best
outcomes. Measure risetime and related properties and compare with the sample interval. 1 Use
the PULSE GENERATOR module set to 10Hz, and various DUTY CYCLES settings to achieve this
easily.

Question 7
For the same settings as in step 17, carry out a quick examination of the frequency response of
the filter. Obtain and record the 3dB cut-off frequency, and the attenuation of the stop-band.

1
You may wish o refer back to your notes from “Experiment 3: Special signals”, where step and
impulse responses were covered.

12-8 © 2011 Emona Instruments Experiment 12 – Sampling and Aliasing V1.2

 As we have already seen, the sinusoid has a special role in linear systems. It turns out that the
sampling properties of sinewaves make it possible to establish precise limits for sampling rates.
This is developed thoroughly in Part 2.

18. Carefully observe the result when the sampling rate is less than two samples per period
(e.g. less than 2kHz). The frequency of the sinewave recovered at the filter output will have
changed (use the filter output as trigger source for the scope). The recovered signal is not the
sinewave at the input of the sampler.

This is easily achieved by slowly decreasing the FREQUENCY setting of the FUNCTION
GENERATOR. Using the “down arrow” to slowly decrement frequency works well.

Check that this outcome occurs with both sampling formats.

One way to see how this comes about is to plot the sample points on graph paper and draw a
smooth curve through these points by eye. The new sinewave is called an alias of the original.
The effect is known as aliasing. Try this below for a sampling rate much less than 2kHz, say
1500Hz. Draw 4 cycles of the 1kHz sinewave as a reference on Graph 1 below.

Graph 1: alias waveforms

Confirm that the sum of the original and alias frequencies = sampling frequency.

Insight into these outcomes is best achieved from a frequency domain perspective.

Experiment 12 – Sampling and Aliasing © 2011 Emona Instruments 12-9

Part 2: through the frequency domain

In Part 1 we examined sampling and reconstitution through observations in the time domain.
However, the mathematical structure underlying these processes is more readily revealed in the
frequency domain.

In the next task we examine the spectrum of the product of two sinewaves. This will be needed
later as a tool for analyzing the spectrum of sampled signals. Patch together some blocks as
shown in Figure 7.

Figure 7:block diagram for product of two sinewaves

19. Set the FUNCTION GENERATOR frequency to 5 kHz. Display the MULTIPLIER output
as well as the lower frequency input sinewave from ANALOG OUTPUT: DAC-1.

Settings are as follows:

FUNCTION GENERATOR: Sine wave output, 2Vpp; 0V offset.
SCOPE: Timebase: 4ms; Ch0: DAC-1 ; Ch1: MULTIPLIER output

The two lines straddling the FUNCTION GENERATOR frequency are clearly displayed. Confirm
that the outcome agrees with the theoretical predictions in Pre-lab preparation Q1. This is an
important fundamental result which you must be familiar and comfortable with.

Next, the spectrum of the sampled sinewave.

20. Keep this patching but change the signal type output from the FUNC OUT terminal of
the FUNCTION GENERATOR to recreate the patching of the sampled sinewave from Figure 2.

Settings are again as follows:

FUNCTION GENERATOR:Squarewave output; 5kHz; 1Vpp, 0.50V Offset, with DUTY
CYCLE=50%
SCOPE: Timebase 4ms; Rising edge trigger on CH0; Trigger level=0V
ANALOG OUTPUT (DAC-1): 4Vpp sinewave at 1kHz

21. Disconnect the scope lead from the MULTIPLIER output and use it to view the sampling
squarewave only. Confirm you understand why its spectrum is a series of odd numbered
harmonics, including a DC component. This was covered in Pre-lab preparation Q2.

22. Reconnect the scope lead to the MULTIPLIER output and view the sampled sinewave
along with the sampling squarewave. Observe the spectral lines straddling each of the
squarewaves harmonics. Each one of these harmonics is straddled by a sum and difference
signal. Satisfy yourself that each of these sinewave components can be considered as being
separately multiplied by the sampler input, i.e. the 1kHz sinewave. Thus, by focusing on just one
Fourier component of the squarewave pulse train in turn, we are able to build the array of line

12-10 © 2011 Emona Instruments Experiment 12 – Sampling and Aliasing V1.2

pairs of the form observed earlier, each pair centered at the respective harmonics of the
FUNCTION GENERATOR squarewave frequency. Confirm you understand completely why the
spectrum looks like this.

Question 8
Explain why the sampled signal spectrum looks the way it does and specifically relate this to
your understanding of pre-lab preparation item 1 & 2.

In Prep Q2 you showed that the squarewave signal can be expressed as a Fourier Series, i.e. as
the weighted sum of sinewaves (in this instance at 5kHz, 15kHz,...) for a 50% duty cycle signal.
Vary the duty cycle of the squarewave by changing the DUTY CYCLE value in the FUNCTION
GENERATOR control panel to 25%.
Satisfy yourself that you understand the source for the appearance now of even harmonics also.

Question 9
Note the frequency of the first and second nulls in the spectrum and explain why they are at
those frequencies.

Experiment with the duty cycle and view the effect of other duty cycles upon the spectrum. Try
using a 10% sampling duty cycle. Confirm with your theoretical understanding.

Aliasing and the Nyquist rate

In the next tasks we revisit the investigation in step 18, where aliasing was discovered at
sampling rates below a critical limit. The view through the frequency domain reveals a
straightforward mathematical interpretation. Once again we will use the FUNCTION
GENERATOR to generate the sampling clock. Whilst we are mainly interested in the frequency
display, in the time domain scope display some clock slippage will be visible but is not of concern.

With the same patching as before in the previous step (Figure 7), select a 25% duty cycle so as
to have both odd and even harmonics present:
Settings are again as follows:
FUNCTION GENERATOR:Squarewave output; 5000 Hz; 1Vpp, 0.50V Offset,
with DUTY CYCLE=25%
SCOPE: Timebase 20ms; Rising edge trigger on CH0; Trigger level=0V
ANALOG OUTPUT (DAC-1): 4Vpp sinewave at 1000Hz. View on CH1

Selecting a time base of 20ms allows us to have higher resolution in the frequency domain and
see the harmonics more closely, although the individual samples in the time domain display are
more difficult to discern.

23. Begin with a sampling rate of 5kHz, which is 5 samples per period of the 1kHz sinewave.
Display the spectrum of the sampled output (on CH1) as well as the spectrum of the sampling

Experiment 12 – Sampling and Aliasing © 2011 Emona Instruments 12-11

pulse train (on CH0). Now progressively reduce the sampling rate in 500Hz steps ie 4500Hz,
4000Hz, 3500Hz, and observe the effect on the spectrum. Carefully keep track of the
positions of the spectrum images about each harmonic of the sampling clock. Note that the only
component that is not shifting is the one at 1000 Hz, corresponding to the input sinewave.
Use RUN /STOP to hold the time domain display if slipping.

24. When you have reduced the sampling rate to 3000Hz, locate the component adjacent to
the one at 1000Hz. It should be at 2000Hz. We denote this component as the lower sideband
of the first spectrum image. (The upper sideband of the first spectrum image is at 4000Hz).
Continue reducing the sample rate and carefully follow the further movement of this component.
Note the sample rate when the lower sideband is at the same frequency as the input sinewave.

Question 10
At what sampling rate does the lower sideband of the first spectrum image become located at
the same frequency as the input sinewave ?

25. Proceed further, to 1500Hz say, and note that the first lower sideband is now at a
frequency below that of the input sinewave. Examine the overall spectrum and check that you
are able to identify the pairing of the sidebands about their respective mirror frequencies.
Note that the image patterns/pairs are now overlapping.

26. Proceed to 1000Hz . Now note that the output also contains a DC component. If you
were sampling at the exact rate of the input signal, and were exactly in phase as well (something
we are not able to due to slippage between the sampling pulses and the input) then you would
expect to sample the sinewave at the exact same point in each period, resulting in a DC output.
Consider this especially in relation to the frequency domain display you have.

Now that we have seen the effect of the sample rate on the spectrum of the sampled sinewave,
we are ready to focus on the recovery of the analog input with a lowpass filter as examined in
step 3 and 4. We will watch the spectrum of the filter input and output as the cut-off
frequency is tuned to suppress the unwanted components, and discover the challenges that arise
as we strive for the lowest achievable sampling rate.

27. Add the TUNEABLE LPF to the experiment setup as shown in Figure 6. Display the
spectrum at the TUNEABLE LOWPASS FILTER output. Set the sampling frequency back up to
5kHz and the TUNEABLE LOWPASS FILTER cut-off to the highest available (Fc knob fully
clockwise). Set the TUNEABLE LPF GAIN to approximately 1 (mid position). Progressively
reduce the TUNEABLE LOWPASS FILTER cut-off so that all components of the sampled
spectrum are suppressed except one. Note its frequency and compare this with the spectrum
of the sampled signal. Satisfy yourself that this is as expected.

Repeat this with the FUNCTION GENERATOR at 1500Hz. Again, compare the result with the
spectrum of the sampled signal and verify the validity of the outcome. Why is the original
analog signal not recovered in this case? Why is the term alias is used to describe this non valid
output?

Question 11

12-12 © 2011 Emona Instruments Experiment 12 – Sampling and Aliasing V1.2

You should be able to recover a clean sinewave. What is its frequency ? Where does it come
from?

In the next segment we look for the lowest sampling rate that allows proper recovery of the
input sinewave without generating alias components.

28. Return to the setup at the start of step 13. Decrease the sampling rate to 2000Hz and
tune the TUNEABLE LOWPASS FILTER as before. Continue reducing the sampling rate and
retuning the TUNEABLE LOWPASS FILTER until the filter can cleanly resolve the desired
input component at 1000Hz, but not reduce its amplitude. We are trying to isolate the wanted
component alone and separate it from the unwanted “images”. Note the sampling rate
corresponding to this situation, and the frequency of the first unwanted sideband. Satisfy
yourself that this frequency corresponds to the lower edge of the stopband of the filter and
that the edge of the passband is at 1000 Hz.

Consider how the sampling rate could be decreased further if a filter with tighter transition
band was available. In theory, an ideal “brickwall” filter is needed to achieve ideal results. We
will have to make do with the capabilities of the TUNEABLE LPF. Check the SIGEx User Manual
to inform yourself about its performance characteristics.

Question 12
Why is it not possible to recover the analog input when the number of samples per cycle of the
input sinewave is less than two?

Question 13
What is the minimum sampling rate that allows a filter to be able to recover the original
sinewave signal without any other unwanted components ?

Multi-frequency input spectrum

Up until now we have simply worked with a single frequency input sinewave for simplicity. The
sampling theorem also works with multiple component input signals such as voice or noise. You
have discovered that the minimum sampling rate that can be used to recover a message
correctly is twice the bandwidth of the signal to be sampled. This rate is commonly known as the
“Nyquist rate”.

In this section we will repeat the previous steps quickly, but with a multi-frequency input signal.
This will allow us to enhance our visualisation of the sampled spectrum and the occurrence of
“aliasing”. There is a two component signal available at ANALOG OUTPUT DAC-0. It has an easy
to recognise “triangular” envelope. This signal is arbitrary and has been created to assist in
visualisation of the sampling process.

Experiment 12 – Sampling and Aliasing © 2011 Emona Instruments 12-13

 Figure 8: example of a 2-component signal at DAC-0: time and frequency representation

Figure 9: example of a sampled multi-frequency signal, in time and frequency domains

29. Repeat the steps 23,24,& 25 in the section above: “Aliasing and the Nyquist rate” whilst
using this multi-frequency signal source. Start with a sampling rate of 10kHz to more clearly see
the images.

Notice how the triangular spectrum is reflected about each harmonic of the sampling signal,
forming a series of spectral pairs.

OPTIONAL: Uses of undersampling in Software Defined Radio

This section is optional and builds upon your understanding of the sampling process.
As has been seen so far, the sampling process can also be thought of as a simultaneous, multi-
frequency multiplying process, where the sampling waveform is a series of harmonics with
decreasing amplitudes. We also know precisely what the outcome of each multiplication will be.

With this in mind, we can take a closer look at the sampling theorem for lowpass signals. Rather
than stating that a signal must be sampled at least at twice the rate of the highest component
in that signal, it states that the signal must be sampled at twice the bandwidth of the signal in
question regardless of its position in the passband. This means that signals in the passband need
only be sampled a minimum of twice the signal’s bandpass bandwidth, and NOT twice the
absolute frequency of the passband signal.

12-14 © 2011 Emona Instruments Experiment 12 – Sampling and Aliasing V1.2

For example, a DSBSC signal of a message with 2Khz bandwidth, at a bandpass frequency
centered at 100kHz (such as is used in EMONA DATEx DSB-SC experiments) need only be
sampled at 2 * 2kHz, and not 2 * 102kHz.

The reason for this is important and has hopefully become evident in previous sections. The
sampling process creates upper and lower sideband pairs about each harmonic of the sampler.
The “Nyquist rate” limitation exists to avoid the created “pairs” overlapping each other and
creating new and unknown components inside the message bandwidth.

This means we can sample at frequencies lower than the passband which is less demanding of
the sampling circuitry and has been utilised in contemporary wireless front-end convertors
which sample directly at RF frequencies, an already demanding task.

In this part of the experiment, we will firstly create a passband DSB-SC signal with a two-tone
“message” signal at a “carrier” frequency of 10kHz. We will then sample this signal at greater
than the 4000Hz (twice the message bandwidth of 2000Hz), but much lower than the actual
maximum frequency of the passband signal itself, 11kHz multiplied by 2 ie: 22kHz.

Remembering that the sampling signal can have harmonics, Fs, 2Fs, 3Fs,… should alert you to the
fact that we are again simply doing a simultaneous, multi-frequency multiplication in which the
smaller 10kHz component (2nd harmonic of 5000Hz) will multiply our message band down to
baseband, where we can directly receive it. This is the essence of a direct down-conversion
receiver in telecommunications.

In effect we are carefully exploiting the “aliasing” effect which in the previous section we were
careful to avoid. Notice that we need to select a sampling frequency which has an integer
harmonic multiple equal to the center of the passband signal.

30. Patch together the experiment setup in Figure 10. The ANALOG OUTPUT: DAC-0 will
supply the passband DSB-SC signal at 10kHz. Select the “PASSBAND” option on the SFP TAB 12
with the toggle switch.

Figure 10: direct down-conversion receiver using undersampling

31. View both inputs to the MULTIPLIER. One is the passband signal, with 2 upper sideband
components, and 2 lower sideband components with a total bandwidth of 2kHz. The other is the
sampling signal with harmonics at DC, 5kHz, 10kHz, 15kHz visible.

Experiment 12 – Sampling and Aliasing © 2011 Emona Instruments 12-15

 Figure 11:examples of inputs and outputs to down-convertor in time and frequency domains

Settings are again as follows:

FUNCTION GENERATOR:Squarewave output; 5kHz; 1Vpp, 0.50V Offset, with DUTY
CYCLE=25%
SCOPE: Timebase 10ms; Rising edge trigger on CH0; Trigger level=0V

32. Remove the scope lead from the sampling signal and use it to view the output of the
MULTIPLIER. Confirm that there is a small image of the “triangular” message from 0 to 1kHz ie
at baseband.

33. View the input and output to the TUNEABLE LPF and gradually tune out all frequencies
except for the baseband message. The characteristic “triangular” envelope of the message in
the frequency domain lets us see that the signal is in fact available at baseband. There will be
some modulation of the signal levels due to non-synchronisation of the sampling signal.
Synchronisation is a major issue in this type of down conversion.

It is also labelled “undersampling” due to the use of low sampling rates, and these techniques
feature in modern Software Defined Radio and wireless systems.

34. Experiment with different sampling rates and confirm that the signals you see are in
accordance with theory. Be aware of the phenonema of negative frequencies “folding” about the
0Hz point and being reflected into the positive frequency domain. This is often the source of
unaccounted-for components.

12-16 © 2011 Emona Instruments Experiment 12 – Sampling and Aliasing V1.2

Tutorial questions

Q1 re need for anti-aliasing filter:

Explain the role of the anti-aliasing filter in the sampling process. Show that
aliasing is caused by the sampling process, hence the anti-aliasing filter must
be ahead of the sampler.
Consider a signal of bandwidth 3.5kHz in the presence of wideband noise. The
noise spectrum is uniform and has a bandwidth of 500kHz. The SNR is 30dB.
The signal is sampled at 8k/s. What is the signal-to-noise-ratio (SNR) of the
sampled signal if a suitable anti-aliasing filter is used. Compare this with the
SNR that would be obtained if an anti-aliasing filter were not used.

Q2 re trade-off: excessively sharp cut-off lowpass filters.

Describe why the use of sharp cut-off anti-aliasing and interpolation lowpass
filters helps achieve sampling rates close to the theoretical limit. Indicate
the disadvantages of sharp cut-off lowpass filters, and the criteria for
obtaining a practical compromise.

Q3 re step 2
Suppose in the model of Fig 1 we were able to increase the duty cycle to the
point of taking up the entire interval between samples, would this be
equivalent to S/Hold ? Describe how you would implement a S/Hold device
(clue: a capacitor is needed to hold the sampled analog voltage).

Q4 Why is S/Hold used in practical DACs, in preference to narrow pulses?

Q5 x/sinx correction
Data sheets of commercial digital to analog converters mention "x/sinx"
correction. Why is this needed?

Experiment 12 – Sampling and Aliasing © 2011 Emona Instruments 12-17

12-18 © 2011 Emona Instruments Experiment 12 – Sampling and Aliasing V1.2