1 Introduction

In the past radio receivers were designed with analog circuitry. This inherently has some problems that all
analog circuits have. That is, they are susceptible to temperature variations, electrical noise, component aging,
and they are complicated and inflexible. Comparing with the analog filters digital filters are far more accurate.
The signal to noise ratio in a DSP radio is higher than the analog radio. Initially, as digital circuits and
processors were developed, they were not useful for radio or any high frequency circuitry since they operated at
low frequencies and their transistor density was not enough for the signal processing needed in receivers.
However, with the exponential increase in transistor density, faster clock rates, and faster A/D converters radio
frequency receivers and possibly higher frequency receivers and transmitters are now suited for the digital
domain. Today, with transistor densities in the billions and clock rates in the GHz range, digital receivers are
everywhere. Because of the advantages of digital communication systems, a concept of Software Defined Radio
(SDR) has become popular in the literature. The ideal concept of SDR is to sample the RF signal with as little as
possible analog manipulation. That is, ideally we would have an A/D converter at the output of an antenna and
do all of the require signal processing in the digital domain. However, sufficiently fast A/D converters are not
cheap enough yet, therefore we still require a front end to generate an intermediate frequency to sample. Once
the signal is in the digital domain the designer has all the benefits of digital signal processing as described
before, and the ease of configuration and reconfiguration.


DSP based FM receiver project is mainly based on Software Defined Radio (SDR) technology. It is a
technology capable of configuring a wireless device to work with any communications system, be it a cellular
phone, a pager, a WI-FI transceiver, an FM or AM radio, a satellite communications terminal and even a garage
door opener. It offers both cost and time savings for consumers, who would only need to buy one radio to meet
multiple communications needs. And more importantly, the same technology can facilitate interoperability
among the communications systems used by military, police and rescue-relief teams, who currently cannot
always communicate with each other, even sometimes in critical, life-threatening situations because of
incompatible radio systems.
This unique radio technology works much like personal computing, where a single hardware platform can carry
out many functions based on the software applications loaded. SDR uses software to perform radio-signal
processing functions instead of using discrete electronic components, or application-specific integrated circuits.
Frequency tuning, filtering, synchronization, encoding and modulation are now functions performed in software
on high-speed reprogrammable devices such as digital signal processors (DSP), field programmable field arrays
(FPGA), or general purpose processors (GPP). RF components are still needed for generation of high
frequencies or for signal amplifications and radiation but SDR aims at reducing their usage to a minimum.









2 Background
Over the last decade as semiconductor technology has improved both in terms of performance capability and
cost, new radio technologies have emerged from military and R&D labs and become mainstream technologies.
One of these technologies is software-defined radio/ DSP radio.
Although much has been discussed in recent years, a good definition of software radio is difficult to generate.
This is largely due to the flexibility that software-defined radios offer, allowing them to take on many different
forms that can be changed to suit the need at hand. However, software-defined radios, or SDRs, do have
characteristics that make them unique in comparison to other types of radios. As the name implies, an SDR is a
radio that has the ability to be transformed through the use of software or redefinable logic. Quite often this is
done with general-purpose digital signal processors (DSPs) or field programmable gate arrays (FPGAs).
In order to take advantage of such digital processing, traditional analog signals must be converted to and from
the digital domain. This is accomplished using analog-to-digital (ADC) and digital-to-analog (DAC) converters.
To take full advantage of digital processing, SDRs keep the signal in the digital domain for as much of the
signal chain as possible, digitizing and reconstructing as close to the antenna as possible, which allows digital
techniques to perform functions traditionally done by analog components as well as others not possible in the
analog domain. There are limits to this, however. Despite the fact that an ADC connected directly to an antenna
is a desirable end goal. However throughout the proposal, I am talking about a FM receiver which do the Digital
signal processing after the IF (Intermediate Frequency) stage.
















3 Aims and Objectives

3.1 Aim of the project

My project aim is to successfully implement a DSP fm receiver. This will involve a hardware/software solution.
The DSP radio can be totally configured or defined by the software so that a common platform can be used
across a number of areas and the software used to change the configuration of the radio for the function required
at a given time. There is also the possibility that it can then be re-configured as upgrades to standards arrive, or
if it is required to meet another role, or if the scope of its operation is changed. Our design will involve some
components associated with the classic superheterodyne receiver; however, a DSP chip (microcontroller) will be
implemented to demodulate the signal. Basically, the detector in the superheterodyne will be replaced with a
DSP solution. Users will be able to receive FM radio frequencies and hear stereo sound via speakers.

3.2 Objectives

Learn Digital Signal Processing (DSP) theory.
Learn communication theory.
Learn Verilog theory.
Learn MATLAB theory regarding filter designing using Fdatool and Filter Builder tool.
Convert MATLAB algorithm into Verilog (HDL coder).
Familiar with the Quartus II integrated development environment (IDE).
Under sampling a band limited signal.
Design low pass, high pass and band pass filters.
Stereo FM de-multiplexing.
Ability to receive FM modulated signal using a common set of hardware.
Ability to alter functionality by downloading and running new software without changing the
hardware.
Recognize and avoid interference with other communication channels.
Generate a clear stereo audio signal.
Get a good knowledge on Digital communication equipments.






4 Approach

First I started my project using STM32F4 Discovery board. This ARM processor (STM32F407VGT6) has fixed
sampling frequency rates due to the pre scalar. Maximum sampling frequency of the STM32F4 processor
(STM32F407VGT6) is 2 MHz and the pre scalar divided frequencies are 2MHz,1MHz,500kHz,250kHz In
order to get the IF signal into base band we need a sampling frequency of 873kHz. In order to achieve the
875kHz sampling frequency and due to very fast parallel processing I choose a FPGA Development board with
in-built Analog to Digital and Digital to Analog converters.

4.1 Work plan

Study Verilog Theory.
Wrote Hello world Verilog codes to light L.E.Ds.
Compiling Verilog codes in Quartus II integrated development environment (IDE).
Study the Analog to Digital and Digital to Analog sample coding which came with FPGA development
board.
Build counters for accurate timing.
Design filters using Matlab Fdatool and Filter Builder.
Convert Matlab code into Verilog code (HDL Coder).
Design accurate filters.
Build a decimation filter.
Build differentiation filter
Zero crossing detection.
PLL Detection.












5 Literature Survey/Theoretical background

5.1 Theory of Frequency Modulation

A general sinusoid is of the form:
( ) u e + = t e
c c
sin
Frequency modulation involves deviating a carrier frequency by some amount. If a sine wave was used to
frequency modulate a carrier, the mathematical expression would be:
t
m c i
e e e e sin A + =
where
frequency modulation
deviation carrier
frequency carrier
frequency ous instantane
=
= A
=
=
m
c
i
e
e
e
e

This expression shows a signal varying sinusoidally about some average frequency. However, we cannot simply
substitute expression in the general equation for a sinusoid. This is because the sine operator acts upon angles,
not frequency. Therefore, we must define the instantaneous frequency in terms of angles.
It should be noted that the amplitude of the modulation signal governs the amount of carrier deviation, while the
modulation frequency governs the rate of carrier deviation.
The term e is an angular velocity and it is related to frequency and angle by the following relationship:
dt
d
f
u
t e = = 2
To find the angle, we must integrate e with respect to time, we obtain:
u e =
}
dt
We can now find the instantaneous angle associated with an instantaneous frequency:
( )
t
f
f
t t t
dt t dt
m
m
c m
m
c
m c i
e e e
e
e
e
e e e e u
cos cos
sin
A
=
A
=
A + = =
} }

This angle can now be substituted into the general carrier signal to define FM:
|
|
.
|

\
| A
= t
f
f
t e
m
m
c fm
e e cos sin
The FM modulation index is defined as the ratio of carrier deviation to modulation frequency:
m
fm
f
f
m
A
=
As a result, the FM equation is generally written as:
( ) t m t e
m fm c fm
e e cos sin =
This is a very complex expression and it is not readily apparent what the sidebands of this signal are like. The
solution to this problem requires knowledge of Bessels functions of the first kind and order p. In open form, it
resembles: