Experiment 22 – Undersampling in SDR (Software Defined Radio)

Preliminary discussion

Software defined radio

A striking feature of the relatively short history of electronic communications is the
significant improvement in performance with each innovation (usually in terms of bandwidth
requirements and/or noise immunity). This has often meant that, as better communications
systems have been introduced, they have quickly replaced existing technologies. For a recent
example of this, consider the switch from analog to digital cell phones.

However, where the existing technology has been too well established to be abandoned, the
new system has run in parallel with the old. For a long-standing example of this, consider the
commercial AM and FM radio systems.

Despite the benefits of new communications techniques, the disadvantages can’t be ignored.
Hardware is either rendered useless or it must be duplicated. These problems have lead to the
development of the latest communications concept called software defined radio (SDR). SDR is
a single tuner that can receive and decode any of the existing communications formats (AM,
FM, DSBSC, ASK, FSK, DSSS, etc). Moreover, it’s is also capable of decoding any
communications format that will be developed in the foreseeable future.

As its name implies, the astounding flexibility of SDR is achieved using software. Instead of
implementing a hardware receiver that is necessarily band and modulation-scheme specific,
SDR is a wideband receiver that converts radio signals to digital then decodes them using the
software appropriate to the modulation scheme of the transmission signal. For a different
modulation scheme, simply change the program. Better still, for a new modulation scheme,
simply install the new program that’s capable of decoding it.

Undersampling
An SDR receiver capable of receiving (and decoding) the majority of electronic communications
would need to operate at frequencies up to and beyond 2.4GHz (a typical cell phone frequency).
Recalling the Nyquist Sample Rate, you might be tempted to imagine the SDR receiver’s
Analog-to-Digital Converter (ADC) needing to sample cell phone signals at over 4.8GHz!
However, the Nyquist requirement to sample at two or more times the highest frequency of
the input signal is for avoiding aliasing of baseband signals.

Bandwidth limited signals (like radio signals in communications) don’t have frequency
components near DC. That being the case, the type of aliasing that the Nyquist Sample Rate
attempts to avoid isn’t a problem. In fact, Shannon’s Information Theorem states that all of
the information in a bandwidth limited signal can be captured with a sampling rate as low as
twice the signal’s bandwidth.

In other words, a 2.4GHz carrier signal with a 30kHz bandwidth can be sampled at a
frequency as low as 60kHz and still capture all of the signal’s information. That said, there are

22-2 © 2007 Emona Instruments Experiment 22 – Undersampling in software defined radio

certain sampling frequencies that will still cause aliasing and there is a mathematical process
for identifying them.

Sampling of bandwidth limited signals at less than the Nyquist Sample Rate is known as
undersampling, band-pass sampling and super-Nyquist sampling. Importantly, as well as allowing
for communications signals up to very high frequencies to be sampled, undersampling has
another significant advantage that makes it ideal for SDR. When the undersampling frequency
is twice the signal’s bandwidth, one of the sampled signal’s aliases occurs at the same
frequency as the original message used to modulate it. In other words, undersampling
demodulates the sampled signal. All that need be done to recover the original message is to
pass it through a low-pass filter to filter out the higher frequency aliases.

The experiment
In this experiment you’ll use the Emona DATEx to set up a bandwidth limited signal then use it
to explore the difference in the spectral composition of a sampled signal produced using a
variety of sampling frequencies above and below the Nyquist Sample Rate. You’ll then use
undersampling to demodulate the bandwidth limited signal and recover the message. Finally,
you’ll explore the effects on the recovered message of mismatches between the modulated
carrier’s bandwidth and the frequency used for undersampling.

It should take you about 40 minutes to complete this experiment.

Equipment

Personal computer with appropriate software installed

NI ELVIS plus connecting leads

NI Data Acquisition unit such as the USB-6251 (or a 20MHz dual channel oscilloscope)

Emona DATEx experimental add-in module

two BNC to 2mm banana-plug leads

assorted 2mm banana-plug patch leads

one set of headphones (stereo)

Experiment 22 – Undersampling in software defined radio © 2007 Emona Instruments 22-3

Part A – Setting up a bandwidth limited signal
To experiment with undersampling you need a bandwidth limited signal. Any of the modulation
schemes can be used for this purpose, but for simplicity of wiring, we’ll use a DSBSC signal.
The first part of the experiment gets you to set one up.

Procedure

1. Ensure that the NI ELVIS power switch at the back of the unit is off.

2. Carefully plug the Emona DATEx experimental add-in module into the NI ELVIS.

3. Set the Control Mode switch on the DATEx module (top right corner) to PC Control.

4. Check that the NI Data Acquisition unit is turned off.

5. Connect the NI ELVIS to the NI Data Acquisition unit (DAQ) and connect that to the
personal computer (PC).

6. Turn on the NI ELVIS power switch at the back then turn on its Prototyping Board
Power switch at the front.

7. Turn on the PC and let it boot-up.

8. Once the boot process is complete, turn on the DAQ then look or listen for the
indication that the PC recognises it.

9. Launch the NI ELVIS software.

10. Launch the DATEx soft front-panel (SFP) and check that you have soft control over the
DATEx board.

11. Launch the NI ELVIS Oscilloscope VI.

12. Set up the scope per the procedure in Experiment 1 ensuring that the Trigger Source
control is set to CH A.

22-4 © 2007 Emona Instruments Experiment 22 – Undersampling in software defined radio

13. Connect the set-up shown in Figure 1 below.

MASTER MULTIPLIER
SIGNALS

DC
X
AC

DC SCOPE
CH A
Y
100kHz AC
SINE
kXY
100kHz
COS MULTIPLIER CH B
100kHz
DIGITAL
8kHz
DIGITAL
X DC TRIGGER
2kHz
DIGITAL
2kHz
SINE
Y DC kXY

Figure 1

This set-up can be represented by the block diagram in Figure 2 below. It generates a 100kHz
carrier that is DSBSC modulated by a 2kHz sinewave message.

Message
To Ch.A
Master Multiplier
Signals module
Y
DSBSC signal
2kHz To Ch.B
X
100kHz
carrier

Master
Signals

Figure 2

14. Adjust the scope’s Timebase control to view two or so cycles of the Master Signals
module’s 2kHz SINE output.

15. Activate the scope’s Channel B input to view the DSBSC signal out of the Multiplier
module as well as the message signal.

Experiment 22 – Undersampling in software defined radio © 2007 Emona Instruments 22-5

16. Set the scope’s Channel A Scale control to the 1V/div position and the Channel B Scale
control to the 2V/div position.

Note: The Multiplier module’s output should be DSBSC signal with alternating halves of
its envelope forming the same shape as the message.

Question 1
For the given inputs to the Multiplier module, what are the frequencies of the two
sinewaves that make up the DSBSC signal?

Question 2
What’s the bandwidth of the DSBSC signal?

17. Suspend the scope VI’s operation by pressing its RUN control once.

18. Launch the NI ELVIS Dynamic Signal Analyzer VI.

19. Adjust the Signal Analyzer’s controls as follows:

General

Sampling to Run

Input Settings

Source Channel to Scope CHB
Voltage Range to ±10V

FFT Settings Averaging

Frequency Span to 150,000
Mode to RMS

Resolution to 400
Weighting to Exponential

Window to 7 Term B-Harris
# of Averages to 3

Triggering

Triggering to Source Channel

Frequency Display

Units to dB
Markers to OFF (for now)

RMS/Peak to RMS

Scale to Auto

22-6 © 2007 Emona Instruments Experiment 22 – Undersampling in software defined radio

20. Verify your answers to Questions 1 and 2 by using the Signal Analyzer’s markers to
determine the frequency of the DSBSC signal’s two sidebands.

Ask the instructor to check

your work before continuing.

Part B – Direct down-conversion using undersampling

If you have successfully completed the experiment on sampling and reconstruction (Experiment
13) you have seen that the mathematical model that defines the sampled signal is:

Sampled signal = the sampling signal × the message

As the sampling signal is a digital signal, the expression can be rewritten as:

Sampled signal = (DC + fundamental + harmonics) × message

When the message signal is modulated carrier like the DSBSC signal that you have set up, the
expression can be rewritten as:

Sampled signal = (DC + fundamental + harmonics) × (LSB + USB)

Solving the expression (which necessarily involves trigonometry that is not shown here) gives:

Duplicates of the LSB and USB (due to their multiplication with sampling signal’s DC
component)

Aliases of the LSB and USB at frequencies equal to the sum and difference of their
frequencies and the sampling signal’s fundamental frequency

Numerous other aliases of the LSB and USB at frequencies equal to the sum and
difference of their frequencies and the sampling signal’s harmonic frequencies

Experiment 22 – Undersampling in software defined radio © 2007 Emona Instruments 22-7

Recall that the math also proves that, where a low-pass filter is being used to reproduce the
original signal by plucking its equivalent out of the sampled signal, the sampling rate must be at
least twice the highest frequency in the original signal. If the sampling rate is less than this,
aliasing occurs.

At first glance then, this suggests that if the DSBSC signal that you have generated is to be
sampled, the sampling rate must be at least 204kHz because of the upper sideband is a
204kHz sinewave.

However, as the DSBSC signal is bandwidth limited (that is, its spectral composition doesn’t
extend down to DC), it’s possible to sample at rates lower than 204kHz without necessarily
causing aliasing. For proof, Table 1 shows some of the aliases produced by sampling the DSBSC
signal at 150kHz.

Table 1
Components due Components due Components due Components due
to DC to fs to 2fs to 3fs
Diff: 48k & 52k Diff: 198k & 202k Diff: 348k & 352k
98k & 102k
Sum: 248k & 252k Sum: 398k & 402k Sum: 548k & 552k

Notice that none of the aliases overlap the 98kHz and 102kHz components in the sampled
signal’s spectral composition. The aliases are either below or above them. So, in this instance,
aliasing wouldn’t occur if a band-pass filter (with sufficiently steep skirts) is used to pluck the
duplicate of the original DSBSC signal out of the sampled signal. That said, aliasing is still
possible by choosing a sampling rate that produces aliases at frequencies that fall inside the
band-pass filter’s pass-band.

Obviously, as the sampling rate decreases, so too do all of the components in the sampled
signal’s spectrum. It makes sense then that, if the right undersampling frequency is used, it
must be possible to produce aliases centre on DC. This is crucial because it means that, when a
modulated carrier is undersampled, one of its sidebands can be directly down-converted back
to a baseband signal without needing to use an intermediate frequency first. All that is needed
is a low-pass filter to reject the other aliases.

A more sophisticated way of understanding direct down-conversion using undersampling

involves thinking of the sampling action as product detection. This is entirely appropriate to do
because the math is almost identical – if you’re not sure about that, compare the notes here
with the notes in the preliminary discussion on product detection in Experiment 9. The
difference is however, instead of multiplying the modulated carrier with a single local
sinusoidal carrier, sampling involves multiplying it with dozens of sinewaves (the sampling
signal’s fundamental and harmonics). Importantly, as long as one of the harmonics is the same
frequency as the modulated carrier, the explanation for a product detector applies equally to
undersampling as a form of demodulation.

22-8 © 2007 Emona Instruments Experiment 22 – Undersampling in software defined radio

To ensure that one of the sampling signal’s harmonics is the same frequency as the modulated
carrier, the sampling rate must be a whole integer sub-multiple of the modulated signal’s
carrier frequency. That said, to avoid aliasing, the sampling rate must be at least twice the
bandwidth limited signal’s bandwidth.

The next part of this experiment lets you demodulate your DSBSC signal to recover the 2kHz
message using undersampling instead of using a product detector.

21. Close the Signal Analyzer’s VI.

22. Restart the scope’s VI by pressing its RUN control once.

23. Return the scope’s Channel B Scale control to the 500mV/div position.

24. Modify the set-up as shown in Figure 3 below.

MASTER MULTIPLIER DUAL ANALOG CHANNEL

SIGNALS SWITCH MODULE
S/ H

DC
X S&H S&H CHANNEL
AC IN OUT BPF

DC SCOPE
CH A
Y IN 1 BASEBAND
1 0 0 kHz AC LPF
SINE
kXY
1 0 0 kHz
COS MULTIPLIER ADDER CH B
CONTROL 1
1 0 0 kHz
DIGITAL CONTROL 2
NOISE
8 kHz
DIGITAL
X DC TRIGGER
2 kHz
DIGITAL
SIGNAL CHANNEL
2 kHz OUT
SINE
Y DC kXY IN 2 OUT

Figure 3

This set-up can be represented by the block diagram in Figure 4 on the next page. The
Multiplier module is used to generate a modulated carrier (DSBSC). The Sample-and-Hold
circuit together with the Baseband LPF is used demodulate it using undersampling.

Experiment 22 – Undersampling in software defined radio © 2007 Emona Instruments 22-9

 Under -sampled
Message DSBSC signal
To Ch.A To Ch.B

Baseband
LPF

Y IN Recovered
S/ H message
2kHz
X CONTROL
100kHz
carrier 8kHz

Master
Signals

DSBSC modulator Demodulation

Figure 4

25. Compare the undersampled DSBSC signal with the original message.

Note: If you look closely, the undersampled DSBSC signal looks a little like an inverted
version of the original message.

26. Modify the scope’s Channel B connection to the set-up as shown in Figure 5 below.

MASTER MULTIPLIER DUAL ANALOG CHANNEL

SIGNALS SWITCH MODULE
S/ H

DC
X S&H S&H CHANNEL
AC IN OUT BPF

DC SCOPE
CH A
Y IN 1 BASEBAND
100kHz AC LPF
SINE
kXY
100kHz
COS MULTIPLIER ADDER CH B
CONTROL 1
100kHz
DIGITAL CONTROL 2
NOISE
8kHz
DIGITAL
X DC TRIGGER
2kHz
DIGITAL
SIGNAL CHANNEL
2kHz OUT
SINE
Y DC kXY IN 2 OUT

Figure 5

22-10 © 2007 Emona Instruments Experiment 22 – Undersampling in software defined radio

 Question 3
What’s the significance of the signal on the Baseband LPF’s output?

Question 4
Given the sampling frequency is 8.333kHz (the signal’s specified value of 8kHz is
rounded down for simplicity), which harmonic in the sampling signal is demodulating the
DSBSC signal?

Ask the instructor to check

your work before continuing.

Experiment 22 – Undersampling in software defined radio © 2007 Emona Instruments 22-11

Part C – Synchronisation
Recall that transmitter and receiver carrier synchronisation is essential to successful
demodulation using product detection. If the local carrier of a product detector has even the
slightest frequency or phase error (relative to the modulated carrier), the demodulated signal
is affected.

Phase errors can reduce the magnitude of the recovered message and even result its complete
cancellation. The effect of frequency errors depends on size. If the error is small (say 0.1Hz)
the message is periodically inaudible but otherwise intelligible. If the frequency error is larger
(say 5Hz) the message is reasonably intelligible but fidelity is poor. When frequency errors are
large, intelligibility is seriously affected. (For a brief explanation of why these effects occur,
refer to Part E in Experiment 9.)

As direct down-conversion using undersampling is a form of product detection, the sampling

signal must be synchronised to the modulated carrier if these effects are to be avoided. The
next part of the experiment let’s you see these effects for yourself.

27. Launch the Function Generator VI.

28. Adjust the Function Generator for an 8.333kHz output.

Note: It’s not necessary to adjust any other controls as the Function Generator’s SYNC
output will be used and this is a digital signal.

29. Disconnect the plug to the Master Signal module’s 8kHz DIGITAL output.

30. Modify the set-up as shown in Figure 6 below.

FUNCTION MASTER MULTIPLIER DUAL ANALOG CHANNEL

GENERATOR SIGNALS SWITCH MODULE
S/ H

DC
X S&H S& H CHANNEL
AC IN OUT BPF

DC SCOPE
ANALOG I/ O CH A
Y IN 1 BASEBAND
100kHz AC LPF
SINE
ACH1 DAC1 kXY
100kHz
COS MULTIPLIER ADDER CH B
CONTROL 1
100kHz
DIGITAL CONTROL 2
ACH0 DAC0 NOISE
8kHz
VARIABLE DC DIGITAL
X DC TRIGGER
+ 2kHz
DIGITAL
SIGNAL CHANNEL
2kHz OUT
SINE
Y DC kXY IN 2 OUT

Figure 6

22-12 © 2007 Emona Instruments Experiment 22 – Undersampling in software defined radio

This modification substitutes the Master Signals module’s 8kHz DIGITAL output for an
8.333kHz digital signal from the Function Generator. This allows you to introduce a phase and
frequency error between the modulated carrier and the “local carrier” (that is, the sampling
frequency’s 12th harmonic).

31. Observe the effect of this change on the recovered message.

Ask the instructor to check

your work before finishing.

Experiment 22 – Undersampling in software defined radio © 2007 Emona Instruments 22-13