Software-Defined Radio (SDR)

For beginners –
By definition and Example
Software-Defined Radio (SDR)
For beginners –
 Content

Introduction…........................................................................................................................ 3

Chapter 1. Background, Theory and Possibilities ...........................................................5

1.1. What is an SDR (Software-Defined Radio) system?............................................5

1.2. GNU Radio Workspace........................................................................................9

1.3. Matlab working environment...........................................................................12

1.4. SDR# Work Environment (SDRSharp)................................................................16

1.5. Some notions about signals and their applications in radio communications...17

1.6. Types of filters and filtering of signals...............................................................20

Chapter 2. SOFTWARE APPLICATIONS FOR SIGNALS ACQUISITION.....................................22

2.1. RTL2832U - R860 SDR Radio Scanner................................................................22

2.2. Instrument control and signal acquisition using SDR#. The influence of the

environment on the reception of signals…………………………………………………………..….23

2.2.1. RTL-SDR Tool Command.............................................................................23

2.2.2. The influence of the environment on the reception of signals...................24

2.3. Applications in GNU Radio using the RTL-SDR system.......................................26

2.3.1. Application 1. Spectrum Analyzer...............................................................26

2.3.2. Application 2. Mono/Stereo FM Receiver..................................................32

 2.3.3. Application 3. AM/NBFM receiver…………………………………………………………..42

2.3.4. Application 4. Receiver/Analyzer with Data Logging..................................50

2.4. Applications for signal analysis, using Matlab...................................................55

2.4.1. Application 5. Processing an audio signal acquired with SDR.....................55

2.4.2. Application 6. Types of modulation............................................................64

Chapter 3. CONCLUSIONS....................................................................................................78

Appendices 80

Appendix 1 – Application code 5......................................................................................80

Appendix 2 – Code related to application 6.....................................................................88

Bibliography....................................................................................................................... 100
Introduction
 Chapter 1. Background, Theory and Possibilities

1.1. What is an SDR (Software-Defined Radio) system?

Software-defined radio (SDR) is a communication system in which components that are

typically implemented using analog hardware—such as mixers, filters, amplifiers, and

modulators/demodulators—are instead realized through software on a computer or

embedded system. Although the concept of SDR has been around for some time,

advancements in digital electronics are making many processes that were once theoretical

become practical.

A basic Software Defined Radio (SDR) system typically consists of a computer equipped with

a sound card or another type of analog-to-digital converter, along with some form of RF front

end. In this design, a significant amount of signal processing is performed by the general-

purpose processor instead of using specialized hardware (electronic circuits). This approach

enables the radio to receive and transmit a wide variety of radio protocols, often referred to

as waveforms, based solely on the software that is being used.

Software-defined radios (SDRs) are highly valuable for military applications and cellular

phone services, as both need to adapt to various changing radio protocols in real time.

Proponents, such as the Wireless Innovation Forum, expect that in the long run, software-

defined radios will become the leading technology in radio communications. SDRs, combined

with software-defined antennas, are essential components that enable cognitive radio.

Traditional receivers have been using the heterodyne pattern for nearly a century, and a

basic understanding of this pattern is necessary to distinguish the difference from the SDR

receiver. The traditional receiver performs three basic functions: setting the carrier
frequency to select the desired signal (tuning), filtering this desired signal against other

received signals, amplifying this signal to compensate for losses on the communication line

Fig. 1 – Block diagram of the heterodyne receiver

The signal received from the antenna is amplified in the RF block which operates only in

the frequencies of a region of interest. Then the signal reaches a mixer where the

frequency of a local oscillator also reaches, through another input. The purpose of the

mixer is to translate the received signal to the intermediate frequency. Usually, the

intermediate frequency coincides with the local oscillator frequency. Further, the signal is

filtered by a band-pass filter, which allows to pass only a specific portion of the spectrum.

Next, the demodulator recovers the original modulated signal.

A typical SDR system comprises an analog front-end and a digital back-end. The front end
deals with the transmission and reception functions of a communication system and contains
analog RF components with tunable parameters, an ADC to convert analog signals, and a
DAC to generate analog signals from digital data. The front-end also uses a device that
performs the conversion for receivers and transmitters, and this device is usually an FPGA.
The front-end part has the main functions of signal conversion, frequency translation to/
from intermediate frequency, power and small signal RF signal amplification with low noise,
automatic gain control [1].
The back-end deals with the digital signal processing. There are also integrated SDR

platforms that combine compute and front-end into a single system.

Fig 2 – Block diagram of a typical basicSDR receiver

In general, an SDR platform consists of several blocks: power supply, clock, reception

and transmission, digital conversion, signal processing. The blocks are connected by

wires that ensure fast data transmission from one block to another.

The structure of an SDR is detailed in the image above. The RF tuner block performs the

same function as the first three blocks on the heterodyne receiver, that of translating the

received signal to the intermediate frequency. Further, the signal is passed through a

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analog-to-digital converter to be digitized. The digital signal reaches the DDC (Digital Down

Converter) block, which contains the mixer, a digital local oscillator and a low-pass filter,

components that work similarly to the classic heterodyne receiver. Finally, the digital signal

reaches the signal processing block (Digital Signal Processing) where operations such as

demodulation, coding and others take place.

The history and development of SDR

 In 1992, Joe Mitola published a paper in which he mentions, for the first time, the

concept of Software Radio. It was created in 1991 by him, being "a radio whose

modulated waves are defined by software";

 In 1998, a company works with SDR to make cellular applications;

 In 2003, Ettus Research offered the first series of USRP – Universal Software

Radio Peripheral;

 In November 2004: FCC approves the first SDR device in the United States, and if

the radio characteristics change, re-approval by the FCC is not required.

 In 2011, the graphical interface for GNU Radio was implemented and Matlab began

to offer support for platforms that use USRP.

 The Advantages of SDR are numerous, among the most important being the following

 Eliminating the hardware component and the related cost, the result being the

simplification of the radio

 architecture and improved performance;

 Adaptable to a wide range of situations, depending on the user's needs;

 The possibility of receiving and transmitting various modulated signals by various

methods, using a common hardware component;

 The ability to choose the frequency and mode of operation most

appropriate to the typical conditions of use;

 Avoiding interference with other communication channels;

 Possibility to modify the functionality by downloading a new improved software

version;

 The opportunity to experiment with various things, such as developing new

protocols.

- Disadvantages of SDR

 The difficulty of writing software for some systems of interest;

 The need to use an interface for digital signals and algorithms;

 Lack of knowledge of users;

 The analog-to-digital converter limits the maximum frequency used by the digital
part of the SDR;
 Developing an SDR requires both software and hardware engineers;

 For some simple radio systems, an SDR platform can be an expensive investment.
The most popular SDR devices in the year 2022 [11]:

1. HackRF One SDR, ANT500 and SMA Antenna Adapter

- Operating frequency: 1MHz – 6GHz;

- SDR half-duplex transceiver;

- Up to 20 million samples per second;

- Compatible with GNU Radio, SDR#

2. Ubertooth One SDR

- 2.4GHz wireless development platform, suitable for experimenting with

the Bluetooth protocol;

- Transmits and receives signals on the 2.4GHz frequency;

- Monitor Bluetooth traffic in real time;

- Open-source platform, hardware and software.

3. YARD Stick One USB Transceiver and 915MHz antenna

- Transmit and receive half-duplex;

- Operating frequencies: 300 – 348MHz, 391 – 464MHz, 782 – 928MHz, but

also operates in other bands unofficially;
 - Modulations: ASK, OOK, GFSK, 2-FSK, 4-FSK, MSK.

The fields of applicability of SDR are very diverse, it can be used in a wide range

of applications, such as [12][13]:

- Radio astronomy, to receive signals from various astronomical objects, such as

stars, galaxies, quasars;

- Digital video broadcasting;

- Research and development, through the ease with which an SDR can be programmed

to transmit or receive information, regardless of the application;

- Military applications, for example the detection and jamming of radar signals from intruders;

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- Communication between military personnel regardless of location or type of equipment used;

- Commercial applications, such as spectrum monitoring, wireless network testing,

interference detection and avoidance;

- Radio amateurs, in applications such as radio communication, satellite tracking,

signal decoding;

- Other applications including spectrum management and interference mitigation.

Fig.3 – Various types of SDR devices

1.2. The GNU Radio desktop

 Since SDR is a type of radio in which the components are implemented in
software, a set of tools is required to process the signal received from the
antenna. GNU Radio is one such toolkit and consists of a package of blocks used
in signal processing, and the implementation of a software defined radio is
achieved by connecting these blocks into a desired structure [2].

GNU Radio is free-to-use software to create SDR applications without the

need for any hardware. GNU Radio uses C++ or Python programming languages in
its implementation [4]. It is a powerful, open source working environment with
flexible analysis and debugging tools, but also a large and capable online
community. It is widely used in research, industry, academia, government
research and the amateur environment.

GNU Radio allows users to create, simulate, and deploy systems that work in

the real world. It is a modular, flowchart-oriented program that

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comes with a rich library of processing blocks, which can be combined to realize complex

signal processing applications [14]. GNU Radio has been used for a wide range of real-

world radio applications such as audio processing, mobile communications, satellite

tracking, radar systems, GSM networks. Thus, the SDR part is moved from analog signal

processing to processing on a computer, using software algorithms.

GNU Radio allows you to use a radio device connected to your computer with
a program written from scratch, but this approach can become cumbersome. For
this reason, GNU Radio provides programs and implementations that are ready to
use. So, GNU Radio implements various functionalities in easy-to-use reusable
blocks, offers excellent scalability, provides a rich library of standard algorithms to
the user, and is optimized for a wide range of platforms. It also provides a variety of
examples to familiarize the user with the work environment [14].
 Fig.4 – The working areas of the GNU Radio programming environment

GNU Radio Companion is a Simulink-like graphical interface that allows the

user to create and run signal processing applications by simply dragging blocks
into the workspace. To extend the functionality of GNU Radio, the user must write
code, and Python is the easiest language to do this. GNU Radio runs on Linux
(recommended), Windows and macOS, on both PC and Raspberry Pi.

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GNU Radio provides the user with over 400 digital signal processing blocks,
including filters, modulators, demodulators, visualization tools, and the option
to add new blocks written in Python or C++.

Allowed connections between blocks:

- From a single block to a single block

- Connection 1 to N outputs

- Multiple entries to a single

block Connections prohibited:

- Loops / feedback

- N to 1 connections
Fig.5 – Allowed and forbidden connections between blocks

Fig.6 – Representation of interconnected blocks

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The blocks:

- They are fundamental signal processing units;

- They are written in C++ or Python;

- They are in arbitrary 1:1, 1:N or N:1 relationships;

- Represents most of the elements that make up a software radio;

- I read certain things, perform processing, write certain

things. Connections:

- They are buffers that allow variables to pass between blocks;

- Memory allocated very efficiently. The

advantages offered by GNU Radio [4]:

- Open source software, available to users of any kind;

- Low cost:

- Can be used in various ways, such as GPS data acquisition, DVD transmitter, data

communications, spectrum measurement;

- It can be integrated not only using SDR, but also using a hardware called USRP

– universal peripheral radio software;

- Easy reconfiguration and the flexibility to set communication parameters when monitoring

radio signals;

- Reconfiguration at the software level, maintaining the same hardware to facilitate the use

of various communication protocols or standards.

1.3. Matlab working environment

Matlab is a programming language and numerical computing environment developed by

Mathworks, which allows manipulation of matrices, graphical representation of functions and data,

implementation of algorithms, creation of graphical interfaces, statistical analysis. Matlab also

contains an additional package called Simulink, used for simulations based on blocks representing

models, and which is used in model-based and integrated or dynamic systems.

Matlab was first commercialized in 1984, although it dates back to the 1960s, having

been invented by mathematician and programmer Cleve Moler. The idea of Matlab was

based on his PhD thesis. By the late 1980s, hundreds of copies of Matlab had been sold

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to universities for student use. The program became popular thanks to the toolboxes

it contained, created by experts in various fields.

A Matlab application is built using the programming language of the same name.

Matlab is frequently used either using the command line or by executing script

files containing Matlab code [21].

Matlab contains a multitude of toolboxes, including:

- Antenna Toolbox;

- Communications System Toolbox;

- DSP System Toolbox;

- Filter Design;

- RF Toolbox;

- Signal Processing Toolbox;

- Risk Management Toolbox and more.

Matlab is very useful in performing simulations because it contains a very diverse collection

of functions that allow the user to implement algorithms from different domains [8]. The

Matlab interface (version R2021a) looks like the figure below:

 Fig.7 – Matlab interface

Matlab is used by millions of engineers and scientists for a wide range of

applications in industry and academia. Such applications include signal processing,

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communications, image and video processing, machine learning, control systems, test

and measurement, and many others [25].

- Creating a graphical interface is done by typing the word in the command

line guides.An interface configuration window like the one in the figure
below will open:

When selecting the optionBlank GUI,an empty interface will open, on which buttons, pop-

up menus, text boxes, labels, graphics can be placed:

Newer versions of Matlab have simplified the process of creating graphical interfaces by making

the App Designer available to the user.

Each graphic element positioned on the interface can be configured by right-clicking

and selecting the Property Inspector.

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The GUI is accompanied by an attached .m file, which contains the predefined functions for

the elements in the interface, but in which the user can also write code to perform various

functions and links between elements, signal processing, graphics display, and more.

Advantages of Matlab [22][23][24]:

- Matlab allows the implementation and testing of algorithms easily;

- Applications with graphical interface can be developed;

- It has a very rich library of predefined functions;

- Has commands to graph signals and work with images;

- It is easy to use and learn and the online community is knowledgeable and

capable;

- It is platform independent and can be installed on various operating systems, such

as Windows, Linux, macOS.

Disadvantages of Matlab [22][23][24]:

- Code in Matlab is more difficult to execute than that in C or C++, because Matlab is

an interpreted language;

- It requires a computer with enough memory and which is fast;

- Relatively expensive to purchase.

15
1.4. SDR# Workbench (SDRSharp)

SDR# is a very popular program used with the RTL-SDR device, being a very easy

program to install and use. Upon opening the program, we see the following interface:

Fig.8 – SDRSharp program interface

SDR# is a simple, intuitive, application-based software for SDR. It is written in

C# with SDR design correctness and performance as goals [6][28].

Devices supported by SDR#:

- SoftRock/Si570;

- HackRF;

- FUNcube Dongle Pro/Pro+;

- RTL-SDR;

- RFSPACE SDR-IQ/-IP/NetSDR;

- IQ file (*.wav).

Some clarifications on the notions present in the graphical interface of SDR#:

NFM=Narrowband Frequency Modulation = typically used by station

radios, weather stations and most VHF/UHF digital signals

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WFM=Wideband Frequency Modulation = type of modulation used by FM radio stations

AM=Amplitude Modulation = used by AM radio stations, frequencies used

by aircraft for voice messages, air traffic control.

LSB/USB=Lower Sideband / Upper Sideband = used by amateurs in the HF band,

to efficiently transmit data and voice with small bandwidths.

CW=Continuous Wave = used to listen to Morse code

DSB=Double Side Band = similar to AM but requires a center frequency. It is not

used very often.

RAW=Raw IQ signal = used to listen or record raw IQ data. Almost never used.

Filters are used to select a signal from the RF window. A good filter will select only

the selected signal. The best option is usually the Blackman-Harris filter.

Filters implemented and selectable in SDR# are: Hamming, Blackman, Blackman-Harris

4, Blackman-Harris 7, Hann-Poisson, Youssef.

I made this short description of the SDR# program because this is the program I used

as a reference in checking the applications I made. In other words, I verified the correctness

and functionality of the applications by checking the reception of certain signals on a certain

frequency and with certain parameters, first using SDR#, to make sure of the certainty of the

reception, and then testing the same with the applications made by me.

1.5. Some notions of signals and their applications in radio communications

 Adigital signalis a signal that represents data in the form of a sequence of discrete
values. At a given time, the signal can have only one value from a finite interval, in
contrast to an analog signal, which is made up of continuous values, and which at a given
time has a real value from a continuous interval of values [ 16].

In signal processing, a digital signal is a representation of an analog signal that has

been sampled and quantized, and the values of the signal only exist at regular time

intervals. So a digital signal can be represented by a sequence of numbers, called

samples. The time interval between two samples drives the sampling frequency.

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Sample rate – sampling frequency–represents the number of values per second that are
sampled from a signal to create a discrete digital signal.

Frequencyis the number of complete oscillations in one second made by a

periodic signal.

Initial phaserepresents the value of the signal phase at the initial time t = 0.

The sampling theorem,formulated by Shannon, says that a band-limited signal, that is,
having a maximum frequency, can be reconstructed from its samples if the sampling
frequency is greater than twice the maximum frequency in its spectrum [9].

Power spectral densityof a signal describes the power present in the signal as
a function of frequency, per unit frequency. Generally, power spectral density is
expressed in W/Hz.

spectrumrepresents the totality of sinusoids from which a signal is formed, and is determined using

the Fourier theorem, in the case of periodic signals, and with the Fourier transform in the case of

non-periodic signals.

Nyquist-Shannon sampling theoremsays that if a function x(t) contains no frequencies

higher than W Hz, then it is completely determined by its samples spaced at least Ts =
1/(2W) seconds. If the sampling theorem is not obeyed, the reconstructed signal will
differ from the original signal sampled at a frequency lower than the Nyquist frequency
[9].
 Fig.9 – Sampling operation

ditheringis the discrete representation of the amplitude of the signals and is an

irreversible process.

Fourier transformhas the role of decomposing a function (a signal) into the frequency

components that make it up. The inverse Fourier transform has the role of recomposing

a function from its direct Fourier transform [9].

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Energy spectral densityis a measure suitable for signals with finite energy (eg pulses)
and shows the distribution of the energy of a signal in frequency [9].

Band of a signalit consists of the frequency domain in which the harmonic

components of a signal are localized. Signals can, in theory, occupy an infinite band,
but to limit this, in practice the notion of a threshold is used, which is an arbitrarily
chosen reference level [9].

Asystemis a device or an algorithm that operates on a signal called input, in

accordance with a set of well-defined rules, to produce as a result another
signal, called the output or response of the system [9].

Amplitude modulationis defined as the operation of multiplying a signal called a

modulator with another signal called a carrier, sinusoidal.

Frequency modulationis defined as the deviation of the instantaneous frequency of the

carrier signal as a function of the instantaneous amplitude of a modulating signal. The
carrier signal can be a sine wave, while the modulating (information) signal can have
any waveform (sine, square, triangle, or other type, such as an audio signal). In
frequency modulation, the amplitude of the carrier signal is always fixed, while the
frequency varies depending on the amplitude of the modulating signal [4].

Analog modulationtranslates the central frequency of the analog signal from the baseband to

the frequency of the carrier signal, and the basic schemes are: AM – amplitude modulation, FM

– frequency modulation, PM – phase modulation [9].

In casedigital modulation,the analog carrier signal is modulated by a discrete signal, this

type of modulation can be considered as a digital-to-analog conversion. This type of

signal transmission is also called digital communication and the basic schemes are: ASK

– Amplitude Shift Keying, FSK – Frequency Shift Keying and PSK – Phase Shift Keying.

Encoding a signalis the data processing operation in the baseband, performed on the basis of

a clock signal, with the frequency equal to that of the bit from the encoder input. The spectrum

of the coded signal is centered on the zero frequency. Binary line codes, with the coded

sequence with two levels, are: NRZ – Non-Return to Zero, BIF – Biphase or Manchester, RZ –

Return to zero, Miller, etc [9].

Aantennais a conductor or system of electrical conductors used to transmit or
receive radio frequency signals. In a communication system, the same antenna
can be used for both transmission and reception [9].

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An isotropic antenna is a theoretical antenna concept in which the emitted radiation
is equal in all three dimensions of space. In real antennas, the directivity
characteristics always differ and the radiation pattern is established following
precise measurements around the antenna [9].

Depending on the distance of the receiver from the transmitter, several areas of interest can be set

in propagation of signals, namely:

- Effective transmission area, where communications can be made and the error rate
is reduced;

- The detection area, where the signal is detected by the receiver and a stable
communication cannot be achieved;

- The interference zone, where the signal may not be detected and contributes additively to
the background noise [9].

Fig.10 – Areas of interest in signal propagation

Attenuation of a signal is the decrease in signal strength, and this increases in proportion to the

distance traveled from the transmitter in the transmission medium.

1.6. Types of filters and signal filtering

Afilterrepresents a functional block that has the property of selecting certain

elements from a set, some being allowed to pass and others being rejected [8]. The
characteristic of a filter is the graphical representation of the behavior of a filter against
a sinusoidal voltage.

Filters can be of several types, depending on their characteristics, the most popular being:

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- Low-Pass Filter: allows signals with a frequency lower than the
selected/set cutoff frequency to pass and attenuates or blocks signals
with a frequency higher than the cutoff frequency;

- High-Pass Filter: allows signals with a frequency higher than the selected/set
cutoff frequency to pass and attenuates or blocks signals with a frequency
lower than the cutoff frequency;

- Band-Pass Filer: allows the passage of frequency signals contained in the

range given by two cutoff frequencies and rejects signals outside this range;

- Band-Stop Filter/Band Reject Filter: allows frequency signals outside a given

range of two cutoff frequencies to pass and rejects signals within this range.

Signal to Noise Ratio (SNR) - The signal-to-noise ratio[4] – represents the ratio of the
desired signal strength to the noise strength and is used as a standard measure of
signal quality for communication systems. An information signal used as a means of
communication will experience interference from noise, interference that can degrade
or even modify the information signal, changing its quality.
21
 Chapter 2. PROCUREMENT SOFTWARE

APPLICATIONS SIGNAL

2.1. RTL2832U SDR Radio Scanner - R860

This SDR system contains Realtek's RTL2832U + R820T2 chipset, a TV

tuner IC manufactured by Rafael Microelectronics in Taiwan.

RTL2832U is a high-performance DVB-T COFDM demodulator that supports USB 2.0 interface.

It is used by radio amateurs, being perhaps the cheapest system of this kind, in the reception

of signals contained in the frequency range 24 MHz - 1.766 GHz. The housing is made of

plastic, which is a weak point compared to other similar but more expensive systems, and the

antenna connector is MCX - female.

 Fig.11 – RTL-SDR system

Primarily, this device was originally designed to be used as a portable tuner to

receive terrestrial television broadcasts (DVB-T), but has evolved and now contains
two distinct functional blocks, namely the tuner and the analog-to-digital conversion
part, digital signal processing and control, and USB connectivity [31].

The block diagram of the device can be found below:

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 Fig.12 – Block diagram of the RTL-SDR system [31]

In the environment of radio amateurs, however, the opportunity to use this device for

hamradio applications was noticed, thus managing to transform this device into an SDR

receiver. Due to the fact that the operating frequency of these devices starts from 20 MHz

upwards, they did not become very popular, being generally used for FM reception.

In newer versions, these devices can receive signals from 20 kHz and allow direct

sampling operation, that is, the tuner block is bypassed and the signal goes directly to

the analog-to-digital converter.

This SDR stick can be used with Windows operating system, several versions, but

also distributions of Linux.

2.2. Instrument control and signal acquisition using SDR#. The influence of the
environment on the reception of signals

2.2.1. RTL-SDR tool command

Controlling the RTL-SDR instrument using the SDR# software is very easy
easy, only the installation of the RTL2832U chipset driver being required at first.
To install this driver, we used a small program called Zadig, available on the
Internet for download [29]. The interface looks like this:

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 Fig.13 – Zadig program interface

Generally, the driver and programs that will be used with the SDR device come with the

device upon purchase.

Connecting to SDR# is very easy, the device just needs to be connected to a USB port

and selected from the list of input devices.

2.2.2. The influence of the environment on the reception of signals

To check the good functionality of the device, but also to highlight

the influence of the environment, namely open space versus closed space on the reception of

signals, I checked the reception of radio stations from the city of Sibiu and from the town of

Zăvideni in Vâlcea county, with the help of the RTL-SDR device, both inside a house and in free

space, using Application 2 of this paper – FM receiver, made in the GNU Radio environment.

To evaluate the quality of the received sound, we used some attributes, which reflect a

subjective assessment, as follows:

- Excellent : reception is achieved without voice distortion, without signal loss

and no interference to disturb the ear;
- Good: signal reception has no interruptions, but noise interference is heard
from time to time;

- environment m: the voice message is received successfully, but the noise is persistent and continuously

received simultaneously with the useful message;

- Poor: the voice message is barely perceptible, drowned in noise;

- Nonexistent : there is no reception on the frequency on which there should be a radio station,

the spectrum does not reveal the presence of energy.

The results of this little experiment can be found in the following tables:

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Table 1 – Signal quality received inside and outside a home, in the city of Sibiu
 Table 2 – The quality of the signal received inside and outside a home, in Zăvideni, at

40 km from Râmnicu Vâlcea

As can be seen, there are some differences between the reception inside a home and the

open space. Even in open space, however, there is the influence of buildings, bridges, which

obstruct and reduce signal quality. Also, one can observe the much better reception of

stations in a city, compared to the reception of radio stations in the countryside, at a

considerable distance from the city.

The reasons why the quality of signal reception differs depending on the environment from which

they are received can be: walls, location in relation to the signal emission source, poor quality of

the equipment (plastic casing), distance from the transmitter.

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2.3. Applications in GNU Radio using the RTL-SDR system

2.3.1. Application 1. Spectrum analyzer

The spectrum analyzer is an instrument similar to the oscilloscope, but which

acquires the data first in the time domain, then calculates the frequency, using the fast

Fourier transform. In this way, it can acquire a large amount of data over a wide

frequency band, calculate amplitude versus frequency, and quickly display it in the

frequency domain. Spectrum analyzers differ from oscilloscopes in that they tend to

introduce background noise (noise floor) lower, and has more special filters that can

differentiate very close signals.

Real-time spectrum analyzers are very useful in capturing very fast

signals, such as those in digital communications. They can generally capture
transient and fast signals faster than sweep analyzers, but they have higher
background noise and a higher price.

The present application is a relatively simple application, introductory to the GNU

Radio work environment, and consists in making a spectrum analyzer, using the RTL-SDR

platform, for the frequency range 22.93 MHz – 1.760 GHz. The GNU Radio environment is

presented in detail in chapter 1.2. of the present work.

The application interface is shown in the figure below:

Fig.14 – Graphical interface of application 1 – window without data recording

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 Fig.15 – Graphical interface of application 1 – data recording window

The functionalities of the application are as follows:

- Visualization of the received spectrum in real time with the help of the RTL-SDR device;

- Real-time visualization of the spectrogram;

- Visualization in the time domain of the received signal;

- Visualization of the type of constellation related to the received signal;

- Selecting the central frequency in the range 22.93MHz – 1.760GHz with the help of a slider, from

the application interface;

- Selecting the bandwidth of the visualized signal, with the help of a slider, from the

application interface;

- Adjusting the gain;

- Adjusting the sampling frequency, in the range of 1Hz – 100MHz;

- Recording of received data in .wav file or other format, with the option to choose the

number of bits for signal acquisition.

In addition to all these functionalities, the Spectrum Analyzer and Time Domain windows allow,

thanks to the GNURadio environment, to adjust several options available in the Control Panel,

accessible by pressing the wheel on the mouse. The same press of the wheel allows adjusting

some parameters in the case of the other two windows, namely Waterfall and Constellation.

27
The block diagram that realizes the previously presented functionalities, realized in GNU Radio:

Fig.16 – Block diagram of application 1

The block diagram contains a multitude of blocks, each with a well-defined function.
variables are represented of boxes without terminals, these being named
Center_frequency, RF_Gain, Bandwidth, sample_rate, bits_per_sample. These are of
chooser or range type to allow the user to adjust their values when using the interface.
In any case, they also have a predefined value that the program runs with at startup.

The RTL-SDR Source block is configured as follows. The configuration is done by double-

clicking on the block:

28
 Fig.17 – Configuration of the RTL-SDR Source block

Sample rate, Frequency, RF Gain and Bandwidth parameters are adjustable from the interface.

The Selector block allows choosing a program execution branch. In this case, we can
select between displaying the signals but without saving data, or with the option to
save data and choose the number of bits to save the data in .wav format. Along with
the Selector block, a QT GUI Chooser block is used, which connects the input to the
Selector to its two outputs. Using the QT GUI Chooser, we choose whether or not to
record data, in the Record data tab.

The QT GUI Tab Widget block is used to arrange the options of the two branches
of the Selector block in two different menus, because when one of the branches is
executed, the other one stops. So I chose to make two menus, one to view the
signals when no data is being recorded, and a second to view the signals while
they are being recorded.

The Throttle block is used to limit the flow of samples so that their average rate
does not exceed the set value. We chose 3.2 MSPS as the limit, which is the
maximum number supported by the RTL-SDR device. Normally, when using
hardware, the Throttle block is not used, with GNU Radio displaying a warning
about this, but it is not contraindicated either, so I chose to put it in the
schematic anyway.

The QT GUI Frequency Sink, QT GUI Waterfall Sink, QT GUI Time Sink and QT GUI

Constellation Sink blocks allow viewing the received signal in multiple forms as seen in

the interface, namely frequency domain, spectrogram, time domain and constellation.

29
The blocks are configured as follows:

- QT GUI Frequency Sink block:

This type of graphical interface allows the display of the power spectral density of the

received signal.

The parameters Center Frequency and Bandwidth are variable, set by the user in

the interface.

GUI Hint represents the coordinates needed to position the display element in the general

interface of the application.

- QT GUI Waterfall Sink block:

30
This type of window displays the spectrogram of one or more summed signals. Multiple

independent signals cannot be displayed simultaneously. Most of the parameters are similar to

those of the Frequency Sink type block.

- QT GUI Time Sink block:

This block displays signals in the time domain, acting as an oscilloscope. Each signal is

represented with a different color. Some configuration options are similar to other blocks.

- Constellation Sink QT GUI block:

31
This block allows the graphical representation of complex data or messages, in the form

of a constellation. A constellation is a tool for displaying the IQ plot of a signal.

Other blocks used in the application block diagram are:

- Complex to Float – data conversion block from complex type to float type. It was

necessary to be able to use the Wav File Sink block, which only allows Float data;

- Wav File Sink – Allows data to be saved in wav format;

- Options – default block of any application in GNURadio, in which the application ID, title,

author, language in which the application's functionality will be compiled (Python or C++)

are set.

Each block in GNURadio has a tab called Generated Code. In this tab you can see the code

generated by the respective block, in the Python language, and the code for the entire application

is generated in a file with the extension .py, after pressing the "Generate the flow graph" button in

the GNURadio interface.

The application made is relatively simple, but it allows the exploration of the spectrum of

the entire range made available by RTL-SDR. If another type of device is desired, it is

sufficient to replace the source block with another block, set the parameters and connect

to the Selector block.

The application allows an analysis of signals and disturbances, but also the acquisition of a

signal in wav format or in any other format, if the Wav File Sink block is replaced, a signal that can

be used in other applications for analysis.

2.3.2. Application 2. Mono/Stereo FM Receiver

An FM receiver is normally an electronic device used to receive radio waves

and convert the information transmitted by them into a usable form, such as
sound. An FM receiver uses an antenna to receive radio signals.
A receiver made with SDR, however, replaces the hardware part of a typical receiver,

using only an antenna connected to an SDR device, controlled by a program, such as the

present application made in GNURadio.

32
 The application made in GNURadio consists of an FM receiver, but with two

demodulation possibilities: mono and stereo. For this, we used two different branches to

implement the two types of receivers, each branch being accessible using the Selector

block and a selection menu in the GUI. The frequency range used by the application is

87 MHz – 108 MHz, as this is the FM band regulated by the authorities in the field, where

most radio stations are found and which use this type of modulation.

Mono versus Stereo sound

The difference between monophonic sound (mono) and stereophonic sound (stereo) is the number

of channels used to record and play the music/sound. Mono signals are recorded and played back

using a single audio channel, while stereo signals use two channels [30]. The audio difference is the

impression of directionality in the case of stereo sound, which gives the impression of being heard

from several directions, while mono sounds give the impression of being heard from only one

direction.

The graphical interface of the application is shown in the following images:

Fig.18 – Graphical interface of application 2

33
Fig.19 – Frequency Domain tab of application 2

Fig.20 – The Time Domain tab of application 2

Fig.21 – Waterfall tab of application 2

34
 Fig.22 – Constellation tab of application 2

The app features are:

- Choice of receiver configuration by the user: mono or stereo;

- FM reception in the 87 MHz – 108 MHz band;

- The options of each type of receiver, to listen without recording, or to listen and record the sound at

the same time;

- The possibility of saving the sound in .wav format, mono or stereo, for
later listening or sound processing;

- Visualization of the raw spectrum received by RTL-SDR, in the Raw

Frequency Representation tab;
- Viewing the spectrum for the mono or stereo receiver, in the Frequency Domain tab, after the

signal processing and filtering stages. The related window is activated when the

desired receiver type is selected;

- The possibility of adjusting the reception frequency, volume and gain in dB,
for better quality reception. A higher gain leads to a reduced noise level;

- Visualization in the time domain of the audio signal, in the Time Domain tab;

- Visualization of the spectrogram of the received raw signal, in the Waterfall tab;

- Viewing the IQ chart, in the Constellation tab.

The block diagram that realizes the previously presented functionalities, realized in GNU Radio:

35
 Fig.23 – Block diagram of application 2

Simplified Block Diagram of Monophonic FM Receiver:

The branch in the schematic in GNURadio, related to the previous block diagram:

Fig.24 – Monophonic FM receiver diagram

- The blockRational Resamplerit has the role of a polyphase FIR filter, which has a frequency at the

input, and at the output the frequency will be that of the input multiplied by interpolation, all

divided by decimation. The interpolation and decimation parameters must have quant values

36
smallest possible. This block performs re-sampling of the signal, before and after filtering.

The first block is configured as follows:

And the second Rational Resampler block in the schematic is configured like this:
- The blockLow Pass Filteris a low-pass filter, where parameters such as sampling

frequency, cutoff and transition window width, in Hz, can be configured. In

GNU Radio, the Low Pass Filter block is configured as follows:

37
 a FIR Type specifies the data types at the input and output of the block; Cutoff Freq is

a given by the cutoff variable, and has the value of 100 kHz; Sample Rate refers to the

a sampling frequency, and has the value of 2 MSPS; The width of the transition band

a (Transition Width) refers to the transition zone between the stop band and the

pass band and is given by the transition variable, which has a value of 1 MHz. The
transition region starts at the cutoff frequency and ends where the filter has
maximum attenuation;

a The Low Pass Filter removes the high frequencies of noise produced during
the demodulation process.

- The blockWBFM Receive(Wideband Frequency Modulation) has the role of

demodulating an FM signal from a broadcast station. Compared to the WBFM
Receive PLL block, it only performs monophonic demodulation with distortion
removal. When entering the WBFM Receive block, the signal is complex, and
when leaving it, the signal is demodulated audio. The configuration of the
block in GNURadio is done in the following way:

a The Quadrature Rate is 500 kHz and the audio decimation factor is
10. The quadrature rate is the sampling frequency of the complex
input signal to the baseband, and the audio decimation factor is the
 factor used to re- sampling the input signal at a frequency to match
the audio's specific sampling frequency.

- The blockMultiply Constantallows adjusting the volume between the values specified by the

QT GUI Range with the identifierbook, which has values between 0 and 1, with a step of

0.001. These values were chosen experimentally.

- The blockAudio Sinkallows a signal to be played, using the speakers of a laptop or

PC. parameterSample Raterefers to the sampling frequency of the audio signal,

and can be selected from a list of the most common frequencies. It's good to know

38
 the fact that not all sample rates will be supported by the user's laptop
or device. For common applications, the frequency of 48 kHz is used.

- The last blockselectit allows the user to listen to the music either without recording

at the same time, or to listen and record at the same time. For the recording

option, the Wav File Sink block is used, configured as follows:

a At File, specify the path of a folder and the name of a .wav file in which to
write the data;

a N Channels refers to the number of audio channels. Being a monophonic signal, the

number of channels is 1.

aAt Sample Rate is specified the sampling frequency used

for sound recording;

a Bits per Sample refers to the number of bits used to save the audio,
per sample.

Simplified Block Diagram of Stereo FM Receiver:

39
The branch in the schematic in GNURadio, related to the previous block diagram:

Fig.25 – Schematic of the Stereophonic FM receiver

- The Rational Resampler block has the same role as the monophonic receiver. The block

is configured as follows:
- The WBFM Receive PLL block has the role of demodulating an FM signal from a

broadcast station. Compared to the WBFM Receive block, it performs full stereo

demodulation. At the input to the WBFM Receive PLL block, the signal is

complex, and at its output, the signal is demodulated audio. The configuration of

the block in GNURadio is done in the following way:

40
 - In addition to these basic blocks for processing the received raw signal, the
application also includes blocks for visualizing the signal, saving the
sound, converting various types of data, as well as blocks with variables.

- The first type blockselectin the schematic, positioned immediately after the RTL-
SDR source, gives the user the ability to choose the type of FM radio receiver
whose related branch to run. This choice can be made using the Receiver
Configuration menu in the interface, from which you can choose Monophonic
FM Receiver or Stereo FM Receiver;

- The blocksQT GUI Frequency Sink: this type of graphical interface allows displaying
the power spectral density of the received signal;

- The blocksQT GUI Time Sink: this block displays signals in the time domain, acting
as an oscilloscope. Each signal is represented with a different color;

- Blocks without terminals represent either variables and their values, or types of menus

to adjust or choose the value of a variable in the interface, during the execution of the

application;

- The blocksFloat to ComplexandComplex to Floatare blocks for converting one data type

to another.

Blocks in GNU Radio have a tab called Generated Code in their composition. There you can view

the code in Python language, generated for that block. Also, the code related to the entire

application can be found in the .py file generated when pressing the "Generate the flow graph"

button in the GNURadio interface:

 The application made is relatively simple, but it allows exploring the FM band between 87

MHz – 108 MHz, adjusting various parameters and recording sound. It also brings the differences

between mono and audio sound to the user, as well as the implementation

41
different in the GNU Radio work environment. If it is desired to use another type of device, it is

sufficient to replace the source block with another block, set the parameters and connect to

the Selector block, the input source selection being planned as further development of the

application.

The application allows an analysis of the spectrum intended for frequency

modulation, but also the acquisition of sound in wav format. The application can be easily

modified by replacing or adding blocks, depending on what the user wants.

The application can be used both by students, in several disciplines that

consider signal processing, both in audio and in the frequency domain, but also by
amateurs, for exploring the FM band or simply listening to the radio when it is not
available internet access.

2.3.3. Application 3. AM/NBFM receiver

AM (Amplitude Modulation)refers to the notion of amplitude modulation, which is a type

of analog modulation and which is defined as the operation of multiplying a signal called a

modulator by another signal called a carrier, sinusoidal. Amplitude modulation is essentially

the modulation of a wave by varying its amplitude, and is particularly used for the purpose of

transmitting an audio signal by combining the modulating signal with a higher frequency

carrier radio signal.

Amplitude modulation is used in applications such as amateur communications,

air traffic control, aircraft communication, radio transceivers, computer modems
(the digital variant of AM – QAM, Quadrature Amplitude Modulation).

Amplitude modulation is typically used in the frequency range 535 kHz – 1,705 MHz [34],

but can also be used at higher frequencies (eg hundreds of MHz) in applications such as

air control.

The schematic of an AM receiver is detailed in the following pages.

 NBFM (Narrowband Frequency Modulation)is a type of frequency
modulation, which, compared to the Wideband Frequency Modulation that is used
by radio stations, is used to transmit voice signals, uses a much smaller portion
than a normal FM channel, and is generally used in wireless communications dus-

42
returned, for example by walkie-talkies or taxi stands. Narrowband systems typically

have lower data rates, while wideband systems support relatively high data rates.

A comparison between wideband FM and narrowband FM can also be found in the table below [33]:

Parameter / Characteristic Wideband FM Narrowband FM

Less than 1 or slightly more

Modulation index Greater than 1

greater than 1

Maximum deviation 75 kHz 5 kHz

Frequency domain

30 Hz – 15 kHz 30 Hz – 3 kHz

Modular

Maximum modulation index 5 – 2500 Just over 1

About 15 times higher Reduced. About the same as

Bandwidth

than that of NBFM at AM

Radio, streaming music to

Applications Police, ambulance, taxi ranks

great quality

Table 3 – Comparison between WBFM and NBFM

To also highlight the differences between amplitude modulation and frequency modulation,

a tabular comparison between these two types of modulation can be found below [35]:

Amplitude modulation Frequency modulation

Frequency and phase remain unchanged Frequency and phase remain unchanged

The modulation index is always more

The modulation index varies between 0 and 1

greater than 1

It has two sidebands in the spectrum It has an infinity of sidebands in the spectrum

The modulator has a simple circuit The modulator has a complex circuit

The amplitude of the carrier signal is The frequency of the carrier signal is changed

modified to transmit the data to transmit the data

The bandwidth is around 10 kHz The bandwidth is around 200 kHz

Frequency range used: 535 kHz – 1,705 Frequency range used: 87 – 108 MHz;

MHz NBFM Uses higher frequencies than that

Poor sound quality Good sound quality

The quality of the received signal is poor The received signal quality is good

Table 4 – Comparison between AM and FM

43
The application made in the GNU Radio work environment consists of an AM/NBFM
receiver, with the possibility to choose the type of receiver whose circuit branch is to
be executed by the application. Also, the supported signal source can be of several
types, as detailed below.

The graphical interface of the application can be found in the following image:

Fig.26 – The graphical interface of the application 3

The app features are:

- Selecting the type of receiver of interest: AM (Amplitude Modulation) or
NBFM (Narrowband Frequency Modulation);

- Adjusting the volume;

- Adjusting the frequency of interest (Center frequency);

- Device gain adjustment (RF Gain);

- Adjusting the displayed frequency band;

- Displaying the received spectrum and the related spectrogram;

- Adjusting the frequency step for the center frequency. In addition to adjusting the step and

sweeping the spectrum with a chosen step, the frequency of interest can be entered

directly into the frequency box on the numeric keypad, then the Enter key is pressed to set

the frequency;

44
 - Adjusting the degree of signal suppression, represented in the diagram by the

Simple Squelch block, valid only for the NBFM receiver;

- Visualization of the spectrum of the received signal;

- Visualization of the spectrogram of the received signal.

In addition to these functionalities made available to the user by the graphical interface, the GNU

Radio environment also offers several options, such as zooming on the graph, saving the spectrum

as an image, setting various parameters such as FFT Size, the minimum and maximum limits of the

graphs, autoscaling and many others .

The block diagram of the application, made in GNU Radio, looks like this:
 Fig.27 – Block diagram of application 3

As can be seen, the general scheme of the application comprises two branches, separated

from the source by means of a type blockselect. This block, with the help of another block

calledQT GUI Chooserwith the identifierchooser, allows the user to choose the receiver type,

and therefore the branch from the selector output to execute. Thus, the upper branch is

intended for the AM receiver, and the lower branch is for the NBFM receiver. In addition to

these blocks are the two display blocks,QT GUI Frequency Sinkand QT GUI Waterfall Sink,

which allow the spectrum and spectrogram of the received signal to be displayed in the

interface using the source.

At the top of the block diagram are the variables, defined with the help of several

types of selectors, such as Range, Chooser, or simply defined as Variable.

45
The source is an Osmocom type, originally developed only for OsmoSDR hardware, and

allows the connection of several types of SDR devices, including:

- RTL2832U sticks based on DBV-T tuners;

- RTL-TCP Spectrum server;

- SDRplay RSP devices;

- AirSpy Wideband receiver;

- Great Scott – HackRF devices;

- RFSPace SDR-IQ, SDR-IT, NetSDR;

- Ettus USRP devices;

- CCCamp 2015;

- Nuand LLC bladeRF;

- Fairwaves XTRX, UmTRX;

- Red Pitaya SDR transceiver;

- FreeSRP.

Only some libraries are required to use these devices via the Osmocom

Source block.
The Osmocom source is configured to support the center frequency starting at 100 kHz,

which the RTL-SDR device used does not cover. In this regard, we have included a

warning to this effect in the interface.

AM receiver block diagram:

Fig.28 – Scheme of the AM receiver

First, the signal from the RTL-SDR source is passed through a low-pass filter to

remove unnecessary information. The filter is configured as follows:

46
The cutoff frequency chosen is 5kHz and the signal is decimated by a decimation factor of 10,

the frequency at the filter output being samp_rate / 10.

The AM demodulator is represented by the blockComplex to Mag,which calculates the magnitude

of complex samples [33].

blocRational Resamplerit has the role of the polyphase FIR filter, which has a frequency at the input,

and at the output the frequency will be that of the input multiplied by interpolation, all divided by

decimation. The interpolation and decimation parameters should be as small as possible. This block

performs re-sampling of the signal after filtering. The block is configured as follows:
blocMultiply Constantallows adjusting the volume between the values specified by the QT GUI

Range with the identifierbook, which has values between 0 and 1, with a step of 0.001. These values

were chosen experimentally.

blocAudio Sinkallows a signal to be played, using the speakers of a laptop or PC. The

block is configured as follows:

47
parameterSample Raterefers to the sampling frequency of the audio signal, and
can be selected from a list of the most common frequencies. Be aware that not all
sample rates will be supported by the user's laptop or device. For common
applications, the frequency of 48 kHz is used.

Block diagram of the NBFM receiver:

Fig.29 – Schematic of the NBFM receiver

First, the signal from the RTL-SDR source is passed through a low-pass filter to

remove unnecessary information. The filter is configured as follows:

48
The cutoff frequency chosen is 10kHz and the signal is decimated by a decimation factor of 10,

the frequency at the output of the filter being samp_rate / 10.

Further, the signal reaches the blockSimple Squelchhas the role of suppressing the

output of the receiver, reducing it to silence, if the signal intensity falls below a certain

level (Threshold). The threshold level in dB can be selected from the interface, by the

user, it is declared as a variable under the name ofsquelchand defaults to -50 dB. The

block is configured as follows:

After this block, the signal reaches the blockNBFM Receive,which demodulates the input signal

and produces at the output a signal with the sampling frequency of 48 kHz, which is the same as

the frequency of the audio signal of interest. The block is configured as follows:
blocMultiply Constantallows adjusting the volume between the values specified by the QT GUI

Range with the identifierbook, which has values between 0 and 1, with a step of 0.001. These values

were chosen experimentally.

blocAudio Sinkallows a signal to be played, using the speakers of a laptop or PC.

Blocks in GNU Radio have a tab called Generated Code in their composition. There you can

view the generated Python code for that block. Also, the code related to the entire application

can be found in the .py file generated when pressing the "Generate the flow graph" button in

the GNURadio interface.

The application allows exploration of different types of AM or NBFM signal sources,

such as ground-to-air transmissions between control towers and aircraft, weather stations,

49
amateurs, taxi radio stations, walkie-talkie signals. It also offers the freedom to

use multiple types of SDR devices via the Osmocom source.

Various blocks can be added to the app to make it more complex depending on the user's

interest. It should be noted that the present application is tested and stable, so including

some blocks or changing parameters in GNU Radio may lead to an unstable version.

Therefore, the application itself, as well as the GNU Radio environment, provide useful

learning tools for students or amateurs interested in these types of modulation.

2.3.4. Application 4. Receiver/Analyzer with Data Logging

The present application is a combination of the first 3 applications, namely: spectrum

analyzer, mono and stereo FM receiver, AM receiver and NBFM receiver, but with some

differences. Each type of receiver has the possibility to save data in wav format, with

selectable parameters.

The source type used is generic,Osmocom Source, compatible with several SDR

devices, including: RTL2832U sticks based on DBV-T tuners, RTL-TCP spectrum server,

SDRplay RSP devices, AirSpy Wideband receiver, Great Scott – HackRF devices,

RFSPace SDR-IQ , SDR-IT, NetSDR, Ettus USRP devices, and the list goes on. It should

be noted that the center frequency can be chosen from the interface in the range of 100

kHz - 6 GHz, but the application was tested only in the range of 22 MHz - 1.766 GHz,

frequencies supported by the RTL-SDR device type tuner, blue stick.

Several types of parameters can be modified from the interface, as I will detail in

the following pages.

The simplified block diagram is represented in the image below:

50
 Fig.30 – Simplified block diagram of application 4

The application interface looks like this:

 Fig.31 – Graphical interface of the application 4

The functionalities of the application are as follows:

- Spectrum exploration in the range 100 kHz – 6 GHz, depending on the type of

device used;

51
- Selection of the type of receiver whose related branch in the block diagram to execute.

The selectable receiver types in the interface are as follows:

- Instrument gain adjustment, from the interface;

- Interface adjustment of volume, center frequency, spectrum sweep step,

displayed bandwidth and parametersquelch, valid only for the

NBFM receiver;

- Real-time visualization of the spectrum, the spectrogram and the received signal, in

the time domain;

- The option to save the signals in wav format, for each type of receiver. The
downside to this option is that you cannot save multiple sounds for the same
receiver, as the file will be overwritten. Also, once the application is closed, the
file must be moved to another location on the computer, or once the
application is opened again, the file will be deleted (empty) so that other new
data can be saved in that file;

- Option to choose sample rate and number of bits per sample for when using
the wav save option. The available sampling frequencies are: 16 kHz, 22.05
kHz, 32 kHz, 44.1 kHz, 48 kHz and the options for the number of bits are 8,
 16, 32 and 64 bits per sample.

The graphical interface is arranged in 3 tabs:

- Spectrum Analyzer without data record–related to the first branch afterselect from the
block diagram. In this window, the spectrum, the spectrogram and the received signal
are visualized, in the time domain, when the selected receiver type is "Spectrum
analyzer without data record", i.e. the first one in the list.

- Spectrum Analyzer with data record–related to the second branch after selectfrom

the block diagram. In this window the spectrum, the spectrogram is visualized

52
 and the received signal, in the time domain, when the selected receiver type is

"Spectrum analyzer with data record", i.e. the second one in the list.

- Different type of receiver –related to the following branches after the blockselect,and in
the windows of this tab are represented the spectrum, spectrogram, signal in the time
domain, for the rest of the receivers in the list (AM, FM, NBFM).

The block diagram made in GNU Radio looks like this:

 Fig.32 – General block diagram of application 4

Fig.33 – General block diagram of application 4 - first half

53
 Fig.34 – General block diagram of application 4 - second half

The blocks used are among those already known:Low Pass Filter, Rational
Resampler, Audio Sink,block types related to receivers, signal display blocks,
variable blocks. The configuration of the blocks is detailed in the previous
applications, which address each type of receiver. A novelty compared to
previous applications is the way we used the blockQT GUI Chooserusing the List
option. This option is valid when choosing an item from a list with more than 5
items. In this case I needed 10 options, and I configured the block like this:
Another novelty is the use of the blockOsmocom Source, which provides more flexibility to the

application, allowing the use of a wide range of SDR devices. The configuration of this block is

done, however, in the same way as the sourceRTL-SDR Source.

54
Blocks in GNU Radio have a tab called Generated Code in their composition. There you can

view the generated Python code for that block. Also, the code related to the entire application

can be found in the .py file generated when pressing the "Generate the flow graph" button in

the GNURadio interface.

The application allows the exploration of the spectrum in the minimum 100 kHz - maximum 6

GHz frequency range, depending on the type of SDR device used, also offering the possibility of

demodulating some frequencies, through the types of implemented receivers. The above application is

distinguished by its user-friendly interface, namely the settings that the user can adjust from the

graphical interface, explicitly named parameters, the possibility of making various adjustments, both

implemented in the program and offered by the GNU Radio development environment.

Various blocks can be added to the app to make it more complex depending on the user's

interest. It should be noted that the present application is tested and stable, so including some

blocks or changing parameters in GNU Radio (in the block diagram and not in the interface)

may lead to an unstable version.

Therefore, the application is a suitable learning tool, suitable for applying the

knowledge learned to various subjects, but also a model for implementing the popular

and common types of receivers, AM, FM and NBFM.

2.4. Signal analysis applications using Matlab

2.4.1. Application 5. Processing an audio signal acquired with SDR

This subchapter presents an application made in the Matlab development environment, an application that

it aims to process an audio signal acquired with the help of an SDR system and the

SDR# program. The audio signal is saved in .wav format and loaded into the interface

where various operations are performed, as will be presented.

Presentation of the graphical interface and functionalities

 The interface was made in the Matlab development environment version 2019a, in the form

of a GUI and a .m script file related to the interface.

The completed interface looks like this:

55
 Fig.35 – The interface for processing an audio signal acquired with SDR

The interface includes 5 spaces for displaying graphics, a series of buttons, text input

boxes and a pop-up menu for selecting the type of filter to apply to the audio signal.

The audio signal is uploaded using the "Upload .wav original file" button,

behind which is the following code sequence:

So, I check the status of the button, and with the help of the functioneye fileI can choose where

to upload the audio file from anywhere on my computer. Next, with the functionaudioread,the

audio signal is read, and returns the data sampled inSound. The frequency at which this

sampling is performed is called Fs.

Code line 190 in the image above converts the loaded two-channel stereo signal to a

mono signal consisting of a single-column matrix, i.e., a single channel. We did this

step so that we could perform various operations further on, such as convolution.

The code related to the interface and the implementation of the functionalities can be found in the

appendices of this work, under the name Dissertatie_Codrina_Lisaru.m. This file contains the functions

56
predefined by Matlab, related to the interface elements, and a function created by

me, namedaudio_signal.

This function includes an initial part, where we declared the variables and read various values of

the interface objects, such as the state of the buttons or the choice of the popup menu items.

Interface components:

- The Upload .wav original file button – allows uploading a .wav format file from anywhere on
the computer;

- The Play original sound button – plays the previously loaded audio signal;

- Original signal spectrum button – displays in the first axis system the spectrum in the
frequency domain of the previously loaded signal;

- The button Save filtered audio – gives the user the option to save the filtered audio signal in a
new .wav file, with the predefined name depending on the type of filter used; The Filter

- Type panel includes the following items:

a List-type pop-up menu: includes the 4 types of filters implemented in the

interface – Pass-Down Filter, Pass-Up Filter, Pass-Band Filter and Stop-Band

a Filter; F cutoff (FTJ, FTS) text box in Hz: allows the user to insert the
cutoff frequency required to filter the signal using the low-pass and
high-pass filters;

a The text boxes Fcut1 and Fcut2 (FTB, FOB) in Hz: allow the user to
insert the two cutoff frequencies necessary to filter the loaded audio
signal, using the two types of filters mentioned in brackets;

aFiltered Signal Spectrum button + Filter Feature – displays the spectrum

 of the filtered audio signal in the second axis system and the ideal and actual

characteristic of the selected filter in the last axis system. In the third and fourth axis

system, the original audio signal in the time domain and the filtered audio signal in

the time domain are displayed, so that you can notice any differences.

a Stop sound button – when pressing this button, the playback of the sound (either
original or filtered) is stopped, the button is very useful because some sounds are of
significant length and the user may not want to play them in full.

- The App info button – when you press it, a new window opens, which contains

the author's name, specialization, and the year she made the app.

57
image of background A the application maybe be found TO address:

https://previews.123rf.com/images/pavalena/pavalena1209/pavalena120900048/15389740-

light-blue-abstract-background-music-notes.jpg

Audio signal filtering. Implementation of filters in Matlab

To example, i have loaded in application a signal . wow

"SDRSharp_20230328_182232Z_101800kHz_AF.wav”, acquired with the help of RTL-SDR

and the SDR# program, signal representing a sequence from a radio broadcast, a man's

voice, the audio signal being affected by radio noise.

After loading the audio signal and pressing the buttons in the interface, the graphs

look like this:

 Fig.36 – Program interface with loaded audio signal

In the second system of axes, the spectrum of the signal can be observed after applying the

band-stop filter on the range 4000Hz – 18760Hz. I chose to leave out the frequencies above

4000Hz because the spectrum of the human voice is sufficiently covered by the components

up to 4000Hz, according to the specialized literature. The result is a voice signal with less

58
noise, but with a slightly reduced quality, due to the removal of high frequency components,

but the message is intelligible and the background noise is significantly reduced.

The last axis system shows, in blue, the ideal characteristic of the chosen filter, and in

red, the actual characteristic. The characteristics and implementation of each type of

filter will be detailed in the following.

Within the code, the implementation of filters is done on the branches of a structureswitch,

which, depending on the type of filter chosen in the interface, uses a certain approach.

To be able to implement filters with finite impulse response (FIR), it is necessary to calculate the coefficients of

the filters, which are calculated according to the relationship:

where N represents the order of the filter. In my program, I used N = 8000.

To implementlow pass filter, we used the following formula to calculate the filter

coefficients:

whereh[n]represent the filter coefficients, which must be calculated.

Thus, in the image below you can find the code sequence in which we implemented the filter, the

graphic representation of the filter characteristic.

59
 Fig.37 – FTJ filter implementation and filter characteristic

functionwheyeyassign ho the matrix of N columns and a single row, which has the value 0 in all

positions. In functionforce,n is calculated according to the formula for the coefficients. function

sync In Matlab it has the value of π built in, so it no longer needs to be written in the formula.

After calculating the coefficients, the amplitude characteristic of the filter is

calculated and the axis on which it will be represented. With the help of the
functionplot, the ideal feature is represented, in blue, and the real feature, in red. The
ideal feature is to draw lines with values of 0 and 1. In reality, this cannot be achieved
because the filters are not causal. A causal system is one where its output depends
only on the input, but in digital signal processing, samples are stored in files and any
of the input values can affect the output signal.
A larger number of coefficients determines a higher quality of the amplitude

characteristic, and therefore of the filtering.

To filter the original signal, I used the functionconv,called convolution, this being the way

to filter signals in the time domain. So, we performed the convolution between two

discrete signals, vizh[n]–filter coefficients, and the variableSound, which represents the

input signal. The relation by which the convolution operation is performed in the case of

finite impulse response filters is:

60
In casehigh pass filter,the formula used to calculate the coefficients is as follows:

The code sequence for implementing the filter, and the graphical representation of the filter

feature can be found in the following image:

 Fig.38 - FTS filter implementation and filter characteristic

For calculating the coefficientsthe bandpass filter, we used the following formula:

61
The code sequence for implementing the filter, and the graphical representation of the filter

feature can be found in the following image:

Fig.39 - FTB filter implementation and filter characteristic

Similarly, the formula for calculating coefficientsof the band-stop filterused is the

following:

Also, the code to implement the filter, and the graphical representation of the

amplitude characteristic of the filter, can be found in the image below:

62
 Fig.40 - FOB filter implementation and filter characteristic

In principle, to implement filters, the steps are the same, as follows:

- Choosing the filter of interest, knowing N – the number of coefficients, the sampling frequency Fs,

the cutting frequency or frequencies, as appropriate;

- The filter coefficients are calculated, according to the formulas above;

- Calculation of the real amplitude characteristic, by applying the Fourier transform of the

filter coefficients and comparing it with the ideal one;

- The use of coefficients for filtering the signal, according to the general formula of the convolution

product between the two signals, namely the signal of interest and the coefficients.

63
2.4.2. Application 6. Types of modulation

The present application aims to support students in learning and studying the

types of modulations used in a multitude of fields. The application is built so that the

most popular types of modulation can be experienced by changing the parameters and

viewing the various types of signals (modulator, carrier, modulated, spectrum). The

application and this chapter are not directly related to SDR systems, but provide a range

of knowledge used both in the acquisition of signals with SDR systems and in the

analysis of these signals and the clearer understanding of the concepts used by analog

and digital data transmission systems .

The types of modulation used in the implementation of the application are those in the scheme

below [36], and next follows a brief description of each type of modulation with examples of

situations in which they are used. Also, the formulas by which we implemented each type of

modulation can be found in the Matlab code related to the application.

Fig.41 – Scheme with the main types of modulation addressed in application 6

 AM – Amplitude Modulation-is a type of analog modulation defined as the operation

of multiplying a signal calledmodulatorwith another signal calledbearer, sinusoidal.

Amplitude modulation is essentially the modulation of a wave by varying its amplitude, and

is particularly used for the purpose of transmitting an audio signal by combining the

modulating signal with a higher frequency carrier radio signal. Amplitude modulation is

used in applications such as amateur communications, air traffic control,

64
communication with aircraft, radio transceivers, modems for computers (digital

variant of AM – QAM, Quadrature Amplitude Modulation) [37].

Fig.42 – Amplitude modulation

By Ivan Akira - Own work, CC BY-SA 3.0,

https://commons.wikimedia.org/w/index.php?curid=9820131

FM – Frequency Modulation–is a modulation technique that consists in

encoding information with the help of a carrier signal, by varying the instantaneous
frequency of this signal. It is used in telecommunications, FM radio, signal
processing and computers. Digital signals can also be transmitted using this
modulation, and this type of frequency modulation is called FSK – frequency-shift
keying. In the narrowband version, NBFM, frequency modulation is used by taxi
stations, ground emergency services, walkie-talkie devices [38].
 Fig.43 – Frequency modulation

By Michel Bakni - Own work, CC BY-SA 4.0, https://

commons.wikimedia.org/w/index.php?curid=68824022

65
 PM – Phase Modulation–is a type of modulation used to condition
communication signals for transmissions. It encodes an information signal in the
form of instantaneous phase variations of a carrier signal. Baseline modulation,
along with frequency modulation, are the main forms of angle modulation. Phase
modulation is commonly used in radio transmission and as an integral part of
many digital transmission coding schemes that underlie technologies such as Wi-
Fi, GSM and satellite television. It is also used in signal and waveform generation
in digital synthesizers [39].

Fig.44 – Phase modulation

By Potasmic - Own work, Public Domain,

https://commons.wikimedia.org/w/index.php?curid=50046376

ASK – Amplitude-Shift Keying–is a type of amplitude modulation that

represents binary data as variations in the amplitude of a signal. Such an ASK
modulated binary signal associates the zero value for low and the carrier
signal value for high, as shown in the image below [9].
Fig.45 – Representation of the ASK signal

66
Applications of this type of modulation include: low frequency RF
applications, home automation devices, tire pressure monitoring systems,
fiber optic data transmission.

FSK – Frequency-Shift Keying–is a type of frequency modulation, where digital

information is encoded using a carrier signal by changing the carrier frequency
between several discrete frequencies, as seen in the image below. This technology is
used in telemetry (the transfer of measured values by a sensor from one point to a
spatially separated location), weather balloon radiosondes, caller ID, garage door
opener systems, low frequency radio transmissions in the very low frequencies (3 –
30 kHz) and extremely low frequencies (3 – 30 Hz). The simplest variant of FSK is
BFSK – binary, in which the frequency changes between two values, associated with
the logical values 1 and 0 [40].

Fig.46 – An example of binary FSK

By No machine-readable author provided. Ktims assumed (based on copyright

 claims). - No machine-readable source provided. Own work assumed (based on

copyright claims)., CC BY-SA 3.0,

https://commons.wikimedia.org/w/index.php?curid=635074

PSK – Phase-Shift Keying–represents the type of digital modulation in which the

phase of the carrier signal is changed by varying the sin and cos inputs at a given time.

Depending on the phases at which the signal changes, there are two types of FSK [40]:

- BPSK – Binary PSK – where the sinusoidal carrier signal has two phase changes (0 and

180 degrees);

67
 Fig.47 – BPSK modulation

- QPSK – Quadrature PSK – in which the sinusoidal carrier signal has four phase

changes (0, 90, 180, 270 degrees), but can be extended to 8 or 16 values [9].

Fig.48 – QPSK modulation

This type of modulation is used in wireless LAN networks, Bluetooth communication, contactless

operations, biometric applications, RFID communication [40]. It is a more efficient type of

modulation in the transmission of radio signals, being more immune to noise and occupying less

bandwidth than other types of modulation.

 QAM – Quadrature Amplitude Modulation–is the name of a family of digital

modulation methods, widely used in modern telecommunications for information

transmission. It is actually a method of combining two modulated either AM or ASK

signals, or two bit streams, into a single channel. Two carrier signals, out of phase by 90

degrees, are modulated and combined. As a result of the 90 degree phase shift, these

signals are in quadrature. One signal is called the in-phase signal, or I, and the other is

the quadrature signal, that is, Q [47][9]. The resulting signal is the combination of the two

signals and has both amplitude and phase variations.

QAM signals are used in data communications, such as digital television,

4G, WiMAX, IEEE 802.11, ADSL [9].

68
 PAM – Pulse-Amplitude Modulation-is a type of modulation to which the message

information is encoded in the amplitude of a series of pulses, so the amplitude of the pulses

is varied. The pulse-modulated signal follows the amplitude of the original signal [42].

Fig.49 – PAM modulation

Uses of PAM include: USB technology – USB 2.0 version uses PAM, some
versions of Ethernet, GDDR6X and GDDR7 based video cards, PCI Express
protocol, digital television [43].
 PWM – Pulse Width Modulation–also known asPulse Duration Modulationis a type of

modulation in which the duration or pulse width of the carrier signal varies in proportion to

the instantaneous amplitude of the information/modulating signal.

The analog method of generating a PWM signal uses a sawtooth signal as the
carrier signal and a sinusoidal signal as the modulating signal. These signals
represent the inputs of a comparator. The result is a PWM signal as seen in
the image below:

69
 Fig.50 – PWM modulation

https://openlabpro.com/wp-content/uploads/2016/10/PWM-generation-analog-circuits.png

The duty cycle of a PWM signal represents what percentage of a period that signal is

in the ON state, i.e. logical '1'. An example is shown below:

Fig.51 – PWM duty factor

 By Thewrightstuff - Own work, CC BY-SA 4.0, https://

commons.wikimedia.org/w/index.php?curid=72876123

PWM signals are often used to regulate the speed of motors, by varying the

duty cycle, dimming, power control in various applications.

PPM – Pulse Position Modulation–is a modulation technique that allows the position of the

pulses to vary depending on the amplitude of the sampled modulating signal. The amplitude and

width of the pulses are constant and only the position of the pulses varies. As the amplitude of the

modulating signal increases, the position of the pulse changes [44].

70
 Fig.52 – Generating the PPM signal

Applications of PPM modulation: military applications, air control, optical

communication systems, remote control of cars, trains, telecommunication systems.

PCM – Pulse-Code Modulation–is a technique by which the analog signal is

converted to digital form for use in digital transmission systems. First, the analog signal is

sampled, then the amplitude of each sample is approximated to the nearest quantization

level. This procedure allows representing the time domain and the amplitude in a discrete

manner [45]. So the major steps to follow in PCM are sampling, quantization, and finally

signal coding, which converts the quantized signal into binary codes. It should be noted

that the coded signal is decoded upon reception to access the original message. The

block diagram of PCM signal generation can be found below [46]:

Fig.53 – Block diagram of PCM generation

71
PCM modulation is used to digitize all forms of analog signal such as video,
music, voice. It is the standard format for digital audio signals for PCs, CDs,
DVDs, but also by digital telephone systems.

Presentation of the graphical interface and functionalities

The application is built in the form of a graphical interface, which allows

viewing different signals and setting certain parameters, interface behind which there
is a .m script file, in which we implemented the modulation types shown above, less
QPSK and QAM. The GUI looks like this:
 Fig.54 – Graphical interface of the application 6

The functionalities of the application are as follows:

- Visualization of signals of interest for each modulation, respectively modulator,

carrier, modulated, bit string where applicable, spectrum of modulated signals;

- Setting parameters of the carrier signal, modulator, modulation index, binary

sequence for digital modulations, number of bits for quantization in the case
of PCM, filling factor of the rectangular signal in the case of PPM;

- Saving the image in png or pdf format, an option provided by the Matlab

development environment;

72
 - Help button (Help!) which provides the user with a short theoretical breviary about each

type of modulation available in the application, and commands that can be used

in Matlab;

- The buttonModulation typeswhich opens a picture with the types of modulation,

depending on the kind of carrier signal (analog or digital) and the resulting data.

Application simulations for each modulation type can be found below:

Fig. 55 – Amplitude Modulation

Fig. 56 – Frequency Modulation

73
Fig. 57 – Phase Modulation
Fig.58 – Amplitude-Shift Keying

Fig.59 – Frequency-Shift Keying

74
Fig.60 – Phase-Shift Keying - Binary
Fig.61 – Pulse-Amplitude Modulation

Fig. 62 – Pulse Width Modulation

75
Fig.63 – Pulse Position Modulation
Fig. 64 – Pulse Code Modulation

Fig.65 – Modulation types button

76
Fig.66 – Help! button
77
 Chapter 3. CONCLUSIONS

The presented applications represent an important source of information and

learning, both for students and for amateurs, having a didactic character. The work
covers both notions of SDR systems and the types of modulation used in everyday
life, signals and terms used in radio communications. The work used the RTL-SDR
device, but the applications are designed to support other types of devices, as
presented in the description of each application. The paper also presents the utility
and ease of use of various software programs, such as GNU Radio and SDR#.

The objectives of this work were:

- familiarization with the working environments necessary to control the tool, such as GNU

Radio, Matlab and SDR#;

- instrument command and signal acquisition using SDR systems;

- analysis of acquired signals using various applications made using Matlab and

GNU Radio;

- creating applications that help students or amateurs to acquire various

knowledge about signals, spectrum, filters;

- highlighting the influence of the propagation environment on a radio signal;

- presentation of the usefulness of the applications made and the possibility of using them in various

situations.

The applications made explore the capabilities of the RTL-SDR device, but they can be extended

and much more varied with the help of other software programs available on the Internet.
All applications have been tested to verify and confirm their functionality under

various conditions.

The applications made are briefly summarized as follows:

- Application 1: Spectrum analyzer – frequency range 22.93 MHz – 1.760 GHz,

real-time view of spectrum, spectrogram, received signal in time domain,

adjustment of bandwidth, gain, sample rate and frequency power stations;

- Application 2: FM Mono/Stereo receiver – selectable mono or stereo configuration,

reception in the 87 – 108 MHz band, sound recording, raw spectrum

visualization, adjustment of frequency, volume, gain in dB;

78
 - Application 3: AM/NBFM receiver – selecting the type of receiver, adjusting the volume,

the center frequency, the displayed frequency band, the gain (Gain), the frequency

step, viewing the spectrum and the spectrogram of the received signal;

- Application 4: Receiver/Analyzer with data logging – a combination of the previous

three applications, with the possibility of using other devices besides the RTL-SDR;

- Application 5: Processing an audio signal acquired with SDR – uploading a file

. wav and filtering it using selectable filters with user-set cut-off

frequency, saving the filtered file to a new file, viewing the signal
spectrum before and after filtering;

- Application 6: Types of modulation – setting the parameters and viewing the signals
that make up the main types of modulation used in communications and
electronics/ computers.

Since the RTL-SDR device only allows the reception of signals, messages can only be

transmitted with another device and with other block schemes, related to a transmitter.

So applications using RTL-SDR only allow reception of AM, FM – Mono, Stereo, NBFM

signal, signal that can be used for signal analysis, audio recordings, audio signal

analysis, filtering, representations in various forms, all these being didactic.

Therefore, the applications made can be used, first of all, in the academic
environment, for learning by testing, but also by amateurs, to explore the
frequency bands received by the RTL-SDR.
79
Appendices

Annex 1 – Code related to application 5

80
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84
85
86
87
Annex 2 – Code related to application 6
88
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103