United International

University
EEE 456
Digital Communication Lab

Group: 04
Experiment-8
Experiment name: Under sampling in SDR
software defined radio

Submitted by:

Md. Tayebur AhmedID: 021 132 007

A.R.M. AtaharIshtiaq ID: 021 142 096
Submitted to:

Arifa Sultana
Date of submission: 11.04.2017

Under sampling in SDR software defined radio

Software-Defined Radio (SDR) is a wireless communications system where all of the
signal processing is mplemented in software. It is an advanced radio technology in
which the modulation and demodulation of radio signals is performed exclusively by
software.

Why is it used?

A communication link consists of three components: the transmitter, the channel,

and the receiver.

To produce a radio that can receive and transmit a new form of radio protocol
just by running new software.
A technique in which all the processing is done in software i.e., mixing,
filtering, demodulation etc.
Used to implement different demodulation scheme and different standards
can be implemented in the same device.
Can be updated so that device doesnt become obsolete with time.

If software does everything what is the role of hardware?

A basic SDR system may consist of a personal computer equipped with a sound
card, or other analog-to-digital converter, preceded by some form of RF front end.
Hardware is still needed for the RF front end and Analog to Digital Convertor (ADC)
and Digital to Analog converter (DAC).

Disadvantages of SDR

The choice of architecture depends on the available technology.

Complexity in using the software.
There are technology limits on achievable RF performances.

Applications Of SDR Technology

Cell phone technology.

Military applications.

Undersampling

In signal processing, under sampling is a technique where one samples a

bandpass-filtered signal at a sample rate below its Nyquist rate (twice the upper
cutoff frequency), but is still able to reconstruct the signal. 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. It is also is known as Undersampling, band-pass
sampling and super-nyquist sampling.

The aliasing effect due to the undersampling technique can be used for our
advantage. When a signal is
sampled at a rate less than twice its maximum frequency, the aliased signal
appears at Fs Fin, where Fs
is the sampling frequency and Fin in the input signal frequency. If we sample the 70-
MHz signal with 100 MSPS sampling rate, the aliased component will appear at 30
MHz (100 70). As we know in advance that the signal is aliased, we can recover the
actual frequency by using the Fs Fin relationship. The under sampling technique
allows the ADC to behave like a mixer or a down converter in the receive chain. For a
band-limited signal of 70 MHz with a 20-MHz signal bandwidth, if the sampling rate
(Fs) is 100 MSPS, the aliased component will appear between 20 MHz to 40 MHz (30
10 MHz).

The experiment

In this experiment you'll use the Emona Telecoms-Trainer 1O1 to set up a bandwidth
limited signal. You'll then use under sampling 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 under sampling.

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

Equipment

. Emona Telecoms-Trainer 101 (plus power-pock)

. Dual channel 20MHz oscilloscope

. two Emona Telecoms-Trainer 101 oscilloscope leads

. assorted Emona Telecoms-Trainer 101 patch leads

Part A - Setting up a bandwidth limited signal

To experiment with under sampling 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 port of the experiment gets you to set one up.
 Figure

Figure Message signal ( blue ) and Mudulated (red)

The Multiplier module's output should be DSBSC signal with alternating halves of its
envelope forming the some 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?

Ans: The frequencies of the two sinewaves that make up the DSBSC signal are
98kHz and 102kHz.

Question 2

What's the bandwidth of the DSBSC signal?

Ans: The bandwidth of the DSBSC signal is 4kHz.

Part B - Direct down-conversion using under sampling

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) x (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 signals
DC component)

- Aliases of the LSB and USB at frequencies equal to the sum and difference of their
frequencies and the sampling signals fundamental frequency

-Numerous other aliases of the LSB and USB at frequencies equal to the sum and
difference of their frequencies and the sampling signals harmonic frequencies

Figure

Figure Message signal (blue) and sample and hold output (red)

Tunable low pass filter output

 Figure Message signal (bue) Demudulated (red)

Question 3

What's the significance of the signal on the Baseband LPFs output?

Ans: The significance of the signal on the Baseband LPFs output is the under-
sampled DSBSC signal looks a little like an inverted version of the original signal.

Question 4

Given the sampling frequency is 8.333kH2 (the signal's specified value of 8kHz is
rounded down for simplicity), which harmonic in the sampling signal is
demodulating the DSBSC signal?

Ans: 13th harmonic in the sampling signal is demodulating the DSBSC signal.

Part C - Synchronization
Recall that transmitter and receiver carrier synchronization 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.

Figure

This modification substitutes the Master Signals module's 8kHz DIGITAL output
 Figure VCO digital output (red) and baseband output (blue)

Figure Baseband low pass flter output (red) Message signal (blue)

The frequency of the digital pulse in S/H control was

We tried to correct the frequency error by turning the VC0 module's Frequency
Adjust control very slightly to the left and right.

Question 5

Why doesn't this solve the problem and allow the demodulator to recover the
message?

Ans: Since it is highly unlikely that the VCO modules output will exactly the right
frequency at exact the right phase for successful demodulation, this doesnt solve
the problem and allow the demodulator to recover the messages.

Discussion
We learned that if a signal is bandlimited we can capture all the information by
sampling it less than Nyquist rate. The Multiplier module's output should be DSBSC
signal with alternating halves of its envelope forming the some shape as the
message.
The Fourier transforms of real-valued functions are symmetrical around the 0 Hz axis. After
sampling, only a periodic summation of the Fourier transform (called discrete-time Fourier
transform) is still available. The individual, frequency-shifted copies of the original transform are
called aliases. The frequency offset between adjacent aliases is the sampling-rate, denoted by fs.
When the aliases are mutually exclusive (spectrally), the original transform and the original
continuous function, or a frequency-shifted version of it (if desired), can be recovered from the
samples. The first and third graphs of Figure 1 depict a baseband spectrum before and after being
sampled at a rate that completely separates the aliases.

The second graph of Figure 1 depicts the frequency profile of a bandpass function occupying the
band (A, A+B) (shaded blue) and its mirror image (shaded beige). The condition for a non-
destructive sample rate is that the aliases of both bands do not overlap when shifted by all integer
multiples of fs. The fourth graph depicts the spectral result of sampling at the same rate as the
baseband function. The rate was chosen by finding the lowest rate that is an integer sub-multiple
of A and also satisfies the baseband Nyquist criterion: fs > 2B. Consequently, the bandpass
function has effectively been converted to baseband. All the other rates that avoid overlap are
given by these more general criteria, where A and A+B are replaced by fL and fH, respectively:[2][3]
The top 2 graphs depict Fourier transforms of 2 different functions that produce the
same results when sampled at a particular rate. The baseband function is sampled
faster than its Nyquist rate, and the bandpass function is undersampled, effectively
converting it to baseband. The lower graphs indicate how identical spectral results
are created by the aliases of the sampling process.