WINTER TRAINING

Project Report
STUDY OF COMMUNICATION SYSTEM
OF ONGC
Submitted by
RAJATSUBHRA KAR
WINTER TRAINEE
Under the guidance of:
Mr. Sukesh Debbarma

Chief Engineer E&T

ONGC Tripura Asset

Department of Electronics & Communication Engineering

National Institute of Technology, Agartala
Tripura-799046

RAJATSUBHRA KAR

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Acknowledgement
I would like to express my deep sense of gratitude and sincere thanks to
my project guide Mr. AnirbanBhattacherjee for his continuous support and
encouragement throughout this period. His meticulous approach in
dealing with complex problem and critical comments at each stage helped
me to make this project a real success. Especially sirs guidance at each
step made me feel as if this period was just a cake walk. He was really
involved with this project and gave his input at all relevant situations. He
allowed me to innovative with my ideas as he never tried to impose his
ideas on me forcibly. I would also like to convey my heartfelt thanks for
offering me this opportunity to undertake this project. Without him it
would not have been possible for me to undertake this project. I would like
to bestow a sincere round of gratitude to him. The hearty support of the
institute: National institute of Technology, Agartala is highly solicited.
The blessings of my loving parents remained with me throughout this
period. Their continuous encouragement made me feel comfortable every
time I ran into a bit of disarray. I shall carry the affection and blessings of
Mr. AnirbanBhattacharjee throughout my life. He has really been an
epitome of hard work, dedication and motivation. His involvement has
really shown me the way to succeed.The list is really unending and there
are many others who helped me through each and every walk of this
project and enabled me to complete it in a soothing and comforting way.
This period has really been quite entertaining and enjoyable. Last but not
least I would like to thank the Almighty for always showering his blessings
on me which enabled me to complete this project.

RAJATSUBHRA KAR

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ABSTRACT

Audio signal processing is an engineering field that focuses on the computational

methods for intentionally altering sounds, methods that are used in many musical
applications.
Here is an overview where everyone can play with audio signals while going deep
into several signal processing topics. We focus on the spectral processing
techniques of relevance for the description and transformation of sounds, developing
the basic theoretical and practical knowledge with which to analyze, synthesize,
transform and describe audio signals in the context of music applications.
The course is based on open software and content. The demonstrations and
programming exercises are done using Matlab, which is a property of mathworks and
the references and materials for the course come from open online repositories and
some digital signal processing books.

RAJATSUBHRA KAR

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CONTENTS
Goal
Objective
Signals
o Introduction
o Audio Signals
o Signal Processing
o Noise
o Additive White Gaussian Noise
o Digital Filtering
o Butterworth Filter
o Wiener Filter
o Audio Signal noise removal using Wiener Filter
o Simulation Result
Conclusion and Future Works
Reference

RAJATSUBHRA KAR

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Goal
The goal of this thesis is to study the nature of audio signals. Analyzing the audio signal in
frequency domain. Study its distinct characteristics and study its behavior under the
application of different digital filters.

Objective
Here our objective is to add an Additive White Gaussian Noise to an audio signal and then
reconstruct the signal using Wiener Filter.

Signals
A signal as referred to in communication systems, signal processing, and electrical
engineering "is a function that conveys information about the behavior or attributes
of some phenomenon". In the physical world, any quantity exhibiting variation in
time or variation in space (such as an image) is potentially a signal that might
provide information on the status of a physical system, or convey a message
between observers, among other possibilities. The IEEE Transactions on Signal
Processing states that the term "signal" includes audio, video, speech, image,
communication, geophysical, sonar, radar, medical and musical signals.
Other examples of signals are the output of a thermocouple, which conveys
temperature information, and the output of a pH meter which conveys acidity
information. Typically, signals are often provided by a sensor, and often the
original form of a signal is converted to another form of energy using a transducer.
For example, a microphone converts an acoustic signal to a voltage waveform, and
a speaker does the reverse.

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Analog and digital signals

Two main types of signals encountered in practice are analog and digital. The
figure shows a digital signal that results from approximating an analog signal by its
values at particular time instants. Digital signals are discrete and quantized, while
analog signals possess neither property.
Digital signals often arise via sampling of analog signals, for example, a
continually fluctuating voltage on a line that can be digitized by an analog-todigital converter circuit, wherein the circuit will read the voltage level on the line,
say, every 50 microseconds and employing a fixed number of bits. The resulting
stream of numbers is stored as digital data on a discrete-time and quantizedamplitude signal. Computers and other digital devices are restricted to discrete
time.

Time discretization
One of the fundamental distinctions between different types of signals is
between continuous and discrete time. In the mathematical abstraction, the domain
of a continuous-time (CT) signal is the set of real numbers (or some interval
thereof), whereas the domain of a discrete-time (DT) signal is the set
of integers (or some interval). What these integers represent depends on the nature
of the signal; most often it is time.

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Discrete-time signal created from a continuous signal by sampling

Digital signal resulting from approximation to an analog signal,

which is a continuous function of time
If for a signal, the quantities are defined only on a discrete set of times, we call it
a discrete-time signal. A simple source for a discrete time signal is the sampling of
a continuous signal, approximating the signal by a sequence of its values at
particular time instants.
A discrete-time real (or complex) signal can be seen as a function from (a subset
of) the set of integers (the index labeling time instants) to the set
of real (or complex) numbers (the function values at those instants).
A continuous-time real (or complex) signal is any real-valued (or complexvalued) function which is defined at every time t in an interval, most commonly an
infinite interval.

Amplitude quantization
If a signal is to be represented as a sequence of numbers, it is impossible to
maintain arbitrarily high precision - each number in the sequence must have a
finite number of digits. As a result, the values of such a signal are restricted to
belong to a finite set; in other words, it is quantized. Quantization is the process of
converting a continuous analog audio signal to a digital signal with discrete
numerical values.

Examples of signals
Signals in nature can be converted to electronic signals by various sensors. Some
examples are:

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Motion: The motion of an object can be considered to be a

signal, and can be monitored by various sensors to provide
electrical signals. For example, radar can provide an
electromagnetic signal for following aircraft motion. A
motion signal is one-dimensional (time), and the range is
generally three-dimensional. Position is thus a 3-vector
signal; position and orientation of a rigid body is a 6-vector
signal. Orientation signals can be generated using
a gyroscope.
Sound: Since a sound is a vibration of a medium (such as
air), a sound signal associates a pressure value to every
value of time and three space coordinates. A sound signal is
converted to an electrical signal by a microphone,
generating a voltage signal as an analog of the sound signal,
making the sound signal available for further signal
processing. Sound signals can be sampled at a discrete set
of time points; for example, compact discs (CDs) contain
discrete signals representing sound, recorded at 44,100
samples per second; each sample contains data for a left
and right channel, which may be considered to be a 2-vector
signal (since CDs are recorded in stereo). The CD encoding is
converted to an electrical signal by reading the information
with a laser, converting the sound signal to an optical signal.
Images: A picture or image consists of a brightness or color
signal, a function of a two-dimensional location. The object's
appearance is presented as an emitted or reflected
electromagnetic wave, one form of electronic signal. It can
be converted to voltage or current waveforms using devices
such as the charge-coupled device. A 2D image can have a
continuous spatial domain, as in a traditional photograph or
painting; or the image can be discretized in space, as in
a raster scanned digital image. Color images are typically
represented as a combination of images in three primary
colors, so that the signal is vector-valued with dimension
three.
Videos: A video signal is a sequence of images. A point in a
video is identified by its two-dimensional position and by the
time at which it occurs, so a video signal has a threedimensional domain. Analog video has one continuous
domain dimension (across a scan line) and two discrete
dimensions (frame and line).
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Biological membrane potentials: The value of the signal is

an electric potential ("voltage"). The domain is more difficult
to establish. Some cells or organelles have the same
membrane potential throughout; neurons generally have
different potentials at different points. These signals have
very low energies, but are enough to make nervous systems
work; they can be measured in aggregate by the techniques
of electrophysiology.

Audio Signals
An audio signal is a representation of sound, typically as an electrical voltage.
Audio signals have frequencies in the audio frequency range of roughly 20 to
20,000 Hz (the limits of human hearing). Audio signals may
be synthesized directly, or may originate at a transducer such as
a microphone, musical instrument pickup, phonograph cartridge, or tape
head. Loudspeakers or headphones convert an electrical audio signal into sound.
Digital representations of audio signals exist in a variety of formats.
An audio channel or audio track is an audio signal communications channel in
a storage device, used in operations such as multi-track recording and sound
reinforcement.

Digital equivalent
As much of the older analog audio equipment has been emulated in digital form,
usually through the development of audio plug-ins for digital audio
workstation (DAW) software, the path of digital information through the DAW (i.e.
from an audio track through a plug-in and out a hardware output) is also called
an audio signal or signal flow.
A digital audio signal being sent through wire can use several formats
including optical (ADAT, TDIF), coaxial (S/PDIF), XLR (AES/EBU),
and Ethernet.

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Audio Signal Processing

Audio signal processing, sometimes referred to as audio processing, is the
intentional alteration of auditory signals, or sound, often through an audio
effect or effects unit. As audio signals may be electronically represented in
either digital or analog format, signal processing may occur in either domain.
Analog processors operate directly on the electrical signal, while digital processors
operate mathematically on the digital representation of that signal.

Audio unprocessed by reverb and delay is metaphorically referred to as "dry",

while processed audio is referred to as "wet".
echo - to simulate the effect of reverberation in a large hall
or cavern, one or several delayed signals are added to the
original signal. To be perceived as echo, the delay has to be
of order 35 milliseconds or above. Short of actually playing a
sound in the desired environment, the effect of echo can be
implemented using either digital or analog methods. Analog
echo effects are implemented using tape
delays and/or spring reverbs. When large numbers of
delayed signals are mixed over several seconds, the
resulting sound has the effect of being presented in a large
room, and it is more commonly
called reverberation or reverb for short.
flanger - to create an unusual sound, a delayed signal is
added to the original signal with a continuously variable
delay (usually smaller than 10 ms). This effect is now done
electronically using DSP, but originally the effect was
created by playing the same recording on two synchronized
tape players, and then mixing the signals together. As long
as the machines were synchronized, the mix would sound
more-or-less normal, but if the operator placed his finger on
the flange of one of the players (hence "flanger"), that
machine would slow down and its signal would fall out-ofphase with its partner, producing a phasing effect. Once the
operator took his finger off, the player would speed up until
its tachometer was back in phase with the master, and as
this happened, the phasing effect would appear to slide up

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the frequency spectrum. This phasing up-and-down the

register can be performed rhythmically.
phaser - another way of creating an unusual sound; the
signal is split, a portion is filtered with an all-pass filter to
produce a phase-shift, and then the unfiltered and filtered
signals are mixed. The phaser effect was originally a simpler
implementation of the flanger effect since delays were
difficult to implement with analog equipment. Phasers are
often used to give a "synthesized" or electronic effect to
natural sounds, such as human speech. The voice of C3PO from Star Wars was created by taking the actor's voice
and treating it with a phaser.
chorus - a delayed signal is added to the original signal with
a constant delay. The delay has to be short in order not to be
perceived as echo, but above 5 ms to be audible. If the delay
is too short, it will destructively interfere with the un-delayed
signal and create a flanging effect. Often, the delayed
signals will be slightly pitch shifted to more realistically
convey the effect of multiple voices.
equalization - different frequency bands are attenuated or
boosted to produce desired spectral characteristics.
Moderate use of equalization (often abbreviated as "EQ")
can be used to "fine-tune" the tone quality of a recording;
extreme use of equalization, such as heavily cutting a
certain frequency can create more unusual effects.
filtering - Equalization is a form of filtering. In the general
sense, frequency ranges can be emphasized or attenuated
using low-pass, high-pass, band-pass or band-stop filters.
Band-pass filtering of voice can simulate the effect of a
telephone because telephones use band-pass filters.
overdrive effects such as the use of a fuzz box can be used
to produce distorted sounds, such as for imitating robotic
voices or to simulate distorted radiotelephone traffic (e.g.,
the radio chatter between starfighter pilots in the science
fiction film Star Wars). The most basic overdrive effect
involves clipping the signal when its absolute value exceeds
a certain threshold.

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pitch shift - this effect shifts a signal up or down in pitch. For

example, a signal may be shifted an octave up or down. This
is usually applied to the entire signal, and not to each note
separately. Blending the original signal with shifted
duplicate(s) can create harmonies from one voice. Another
application of pitch shifting is pitch correction. Here a
musical signal is tuned to the correct pitch using digital
signal processing techniques. This effect is ubiquitous in
karaoke machines and is often used to assist pop singers
who sing out of tune. It is also used intentionally for
aesthetic effect in such pop songs
as Cher's Believe and Madonna's Die Another Day.
time stretching - the complement of pitch shift, that is, the
process of changing the speed of an audio signal without
affecting its pitch.
resonators - emphasize harmonic frequency content on
specified frequencies. These may be created from
parametric EQs or from delay-based comb-filters.
robotic voice effects are used to make an actor's voice
sound like a synthesized human voice.
synthesizer - generate artificially almost any sound by either
imitating natural sounds or creating completely new sounds.
modulation - to change the frequency or amplitude of a
carrier signal in relation to a predefined signal. Ring
modulation, also known as amplitude modulation, is an
effect made famous by Doctor Who'sDaleks and commonly
used throughout sci-fi.
compression - the reduction of the dynamic range of a sound
to avoid unintentional fluctuation in the dynamics. Level
compression is not to be confused with audio data
compression, where the amount of data is reduced without
affecting the amplitude of the sound it represents.
3D audio effects - place sounds outside the stereo basis
reverse echo - a swelling effect created by reversing an
audio signal and recording echo and/or delay while the
signal runs in reverse. When played back forward the last
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echos are heard before the effected sound creating a rush

like swell preceding and during playback. Jimmy Page of Led
Zeppelin used this effect in the bridge of "Whole Lotta Love".
active noise control- a method for reducing unwanted sound
wave field synthesis - a spatial audio rendering technique for
the creation of virtual acoustic environments

Noise
In electronics, noise is a random fluctuation in an electrical signal, a characteristic
of all electronic circuits.Noise generated by electronic devices varies greatly, as it
can be produced by several different effects. Thermal noise is unavoidable at nonzero temperature (see fluctuation-dissipation theorem), while other types depend
mostly on device type (such as shot noise, which needs steep potential barrier) or
manufacturing quality and semiconductor defects, such as conductance
fluctuations, including 1/f noise.
In communication systems, noise is an error or undesired random disturbance of a
useful information signal in a communication channel. The noise is a summation of
unwanted or disturbing energy from natural and sometimes man-made sources.
Noise is, however, typically distinguished from interference, (e.g. cross-talk,
deliberate jamming or other unwanted electromagnetic interference from specific
transmitters), for example in the signal-to-noise ratio (SNR), signal-to-interference
ratio (SIR) and signal-to-noise plus interference ratio (SNIR) measures. Noise is
also typically distinguished from distortion, which is an unwanted systematic
alteration of the signal waveform by the communication equipment, for example in
the signal-to-noise and distortion ratio (SINAD). In a carrier-modulated passband
analog communication system, a certain carrier-to-noise ratio (CNR) at the radio
receiver input would result in a certain signal-to-noise ratio in the detected
message signal. In a digital communications system, a certain Eb/N0 (normalized
signal-to-noise ratio) would result in a certain bit error rate (BER).
While noise is generally unwanted, it can serve a useful purpose in some
applications, such as random number generation or dithering.

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Analog display of random fluctuations in

voltage: e.g., pink noise.

ADDITIVE WHITE
GAUSSIAN NOISE
Additive white Gaussian noise (AWGN) is a basic noise model used
in Information theory to mimic the effect of many random processes that occur in
nature. The modifiers denote specific characteristics:
Additive because it is added to any noise that might be
intrinsic to the information system.
White refers to the idea that it has uniform power across
the frequency band for the information system. It is an
analogy to the color white which has uniform emissions at all
frequencies in the visible spectrum.
Gaussian because it has a normal distribution in the time
domain with an average time domain value of zero.
Wideband noise comes from many natural sources, such as the thermal vibrations
of atoms in conductors (referred to as thermal noise or Johnson-Nyquist
noise), shot noise, black body radiation from the earth and other warm objects, and
from celestial sources such as the Sun. The central limit theorem of probability
theory indicates that the summation of many random processes will tend to have
distribution called Gaussian or Normal.
AWGN is often used as a channel model in which the only impairment to
communication is a linear addition of wideband or white noise with a
constant spectral density (expressed as watts per hertz of bandwidth) and
a Gaussian distribution of amplitude. The model does not account
for fading, frequency selectivity, interference, nonlinearity or dispersion. However,
it produces simple and tractable mathematical models which are useful for gaining
insight into the underlying behavior of a system before these other phenomena are
considered.
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The AWGN channel is a good model for many satellite and deep space
communication links. It is not a good model for most terrestrial links because of
multipath, terrain blocking, interference, etc. However, for terrestrial path
modeling, AWGN is commonly used to simulate background noise of the channel
under study, in addition to multipath, terrain blocking, interference, ground clutter
and self-interference that modern radio systems encounter in terrestrial operation.

Digital Filtering

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In signal processing, a digital filter is a system that performs mathematical

operations on a sampled, discrete-time signal to reduce or enhance certain aspects
of that signal. This is in contrast to the other major type of electronic filter,
the analog filter, which is an electronic circuit operating on continuous-time analog
signals.
A digital filter system usually consists of an analog-to-digital converter to sample
the input signal, followed by a microprocessor and some peripheral components
such as memory to store data and filter coefficients etc. Finally a digital-to-analog
converter to complete the output stage. Program Instructions (software) running on
the microprocessor implement the digital filter by performing the necessary
mathematical operations on the numbers received from the ADC. In some high
performance applications, an FPGA or ASIC is used instead of a general purpose
microprocessor, or a specialized DSP with specific paralleled architecture for
expediting operations such as filtering.
Digital filters may be more expensive than an equivalent analog filter due to their
increased complexity, but they make practical many designs that are impractical or
impossible as analog filters. When used in the context of real-time analog systems,
digital filters sometimes have problematic latency (the difference in time between
the input and the response) due to the associated analog-to-digital and digital-toanalog conversions and anti-aliasing filters, or due to other delays in their
implementation.
Digital filters are commonplace and an essential element of everyday electronics
such as radios, cellphones, and AV receivers.
A digital filter is characterized by its transfer function, or equivalently,
its difference equation. Mathematical analysis of the transfer function can describe
how it will respond to any input. As such, designing a filter consists of developing
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specifications appropriate to the problem (for example, a second-order low pass

filter with a specific cut-off frequency), and then producing a transfer function
which meets the specifications.
The transfer function for a linear, time-invariant, digital filter can be expressed as a
transfer function in the Z-domain; if it is causal, then it has the form:

Where, the order of the filter is the greater of N or M.

Butterworth Filter

The frequency response plot from Butterworth's 1930 paper.

The Butterworth filter is a type of signal processing filter designed to have as flat
a frequency response as possible in the passband. It is also referred to as
a maximally flat magnitude filter. It was first described in 1930 by the
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British engineer and physicist Stephen Butterworth in his paper entitled "On the
Theory of Filter Amplifiers"

Matlab code of using Butterworth filter in bandpass

mode

Fs=16384;
Order=2;
Sampling_freq=16384;
[B,A]=butter(Order,[0.7,0.9]);
freqz(B,A,5000,Sampling_freq);
[Flute,Fs]=wavread('FluteMixRingtone.wav');
P=filter(B,A,Flute);
soundsc(P,Fs)
Q=fft(P,4096);
Pyy=Q.*conj(Q)/4096;
f_val=3e4/4096*(0:2048);
plot(f_val,Pyy(1:2049))

Original Singal (amplitude vs frequency)

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Filter (Butterworth Bandpass)

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Filtered Signal(Amplitude vs Frequency)

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NOISE REMOVAL
Noise reduction is the process of removing noise from a signal.
All recording devices, both analog or digital, have traits which make them
susceptible to noise. Noise can be random or white noise with no coherence, or
coherent noise introduced by the device's mechanism or processing algorithms.
In electronic recording devices, a major form of noise is hiss caused by
random electrons that, heavily influenced by heat, stray from their designated path.
These stray electrons influence the voltage of the output signal and thus create
detectable noise.

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In the case of photographic film and magnetic tape, noise (both visible and
audible) is introduced due to the grain structure of the medium. In photographic
film, the size of the grains in the film determines the film's sensitivity, more
sensitive film having larger sized grains. In magnetic tape, the larger the grains of
the magnetic particles (usually ferric oxide or magnetite), the more prone the
medium is to noise.
To compensate for this, larger areas of film or magnetic tape may be used to lower
the noise to an acceptable level.
In selecting a noise reduction algorithm, one must weigh several factors:
the available computer power and time available: a digital camera must
apply noise reduction in a fraction of a second using a tiny onboard CPU,
while a desktop computer has much more power and time
whether sacrificing some real detail is acceptable if it allows more noise to
be removed (how aggressively to decide whether variations in the image are
noise or not)
the characteristics of the noise and the detail in the image, to better make
those decisions

Wiener Filter
In signal processing, the Wiener filter is a filter used to produce an estimate of
a desired or target random process by linear time-invariant (LTI) filtering of an
observed noisy process, assuming known stationary signal and noise spectra,
and additive noise. The Wiener filter minimizes the mean square error between
the estimated random process and the desired process.

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The goal of the Wiener filter is to compute a statistical estimate of an unknown

signal using a related signal as an input and filtering that known signal to produce
the estimate as an output. For example, the known signal might consist of an
unknown signal of interest that has been corrupted by additive noise. The Wiener
filter can be used to filter out the noise from the corrupted signal to provide an
estimate of the underlying signal of interest. The Wiener filter is based on
a statistical approach, and a more statistical account of the theory is given in
the minimum mean-square error (MMSE) article.
Typical deterministic filters are designed for a desired frequency response.
However, the design of the Wiener filter takes a different approach. One is
assumed to have knowledge of the spectral properties of the original signal and the
noise, and one seeks the linear time-invariant filter whose output would come as
close to the original signal as possible. Wiener filters are characterized by the
following:[1]
1. Assumption: signal and (additive) noise are stationary
linear stochastic processes with known spectral
characteristics or known autocorrelation and crosscorrelation
2. Requirement: the filter must be physically
realizable/causal (this requirement can be dropped, resulting
in a non-causal solution)
3. Performance criterion: minimum mean-square error (MMSE)
This filter is frequently used in the process of deconvolution; for this application,
see Wiener deconvolution.

IMPLEMENTATION OF WIENER FILTER

The inverse filtering is a restoration technique for deconvolution, i.e., when
the image is blurred by a known lowpass filter, it is possible to recover the
image by inverse filtering or generalized inverse filtering. However, inverse
filtering is very sensitive to additive noise. The approach of reducing one
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degradation at a time allows us to develop a restoration algorithm for each

type of degradation and simply combine them. The Wiener filtering executes
an optimal tradeoff between inverse filtering and noise smoothing. It
removes the additive noise and inverts the blurring simultaneously.
The Wiener filtering is optimal in terms of the mean square error. In other
words, it minimizes the overall mean square error in the process of inverse
filtering and noise smoothing. The Wiener filtering is a linear estimation of
the original image. The approach is based on a stochastic framework. The
orthogonality principle implies that the Wiener filter in Fourier domain can
be expressed as follows:

where
are respectively power spectra of the
original image and the additive noise, and H(f1,f2) is the
blurring filter. It is easy to see that the Wiener filter has two
separate part, an inverse filtering part and a noise
smoothing part. It not only performs the deconvolution by
inverse filtering (highpass filtering) but also removes the
noise with a compression operation (lowpass filtering).
Implementation

To implement the Wiener filter in practice we have to

estimate the power spectra of the original image and the
additive noise. For white additive noise the power spectrum
is equal to the variance of the noise. To estimate the power
spectrum of the original image many methods can be used.
A direct estimate is the periodogram estimate of the power
spectrum computed from the observation:

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where Y(k,l) is the DFT of the observation. The advantage of

the estimate is that it can be implemented very easily
without worrying about the singularity of the inverse
filtering. Another estimate which leads to a cascade
implementation of the inverse filtering and the noise
smoothing is

which is a straightforward result of the fact:

The
power spectrum Syycan be estimated directly from the
observation using the periodogram estimate. This estimate
results in a cascade implementation of inverse filtering and
noise smoothing:

The disadvantage of this implementation is that when the

inverse filter is singular, we have to use the generalized
inverse filtering. People also suggest the power spectrum of
the original image can be estimated based on a model such
as the
model.

Wiener Filter function creation in

MATLAB
function ex = wienerFilter(y,h,sigma,gamma,alpha)
N = size(y,1);
Yf = fft2(y);
Hf = fft2(h,N,N);
Pyf = abs(Yf).^2/N^2;
sHf = Hf.*(abs(Hf)>0)+1/gamma*(abs(Hf)==0);
iHf = 1./sHf;
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iHf =
iHf.*(abs(Hf)*gamma>1)+gamma*abs(sHf).*iHf.*(abs(sHf)*gamma<=1
);
Pyf = Pyf.*(Pyf>sigma^2)+sigma^2*(Pyf<=sigma^2);
Gf = iHf.*(Pyf-sigma^2)./(Pyf-(1-alpha)*sigma^2);
eXf = Gf.*Yf;
ex = real(ifft2(eXf));
return

Addition of AWGN to an audio file and

reconstruction using Wiener Filtering:
closeall;
clearall;
Fs=4096;
[A,Fs]=wavread('Aguner.wav');
R=A(round(1/2*end:end));
SNR=1; %dB
P=awgn(R,SNR,'measured');
figure(1), plot(P), hold, plot(R,'r'),
xlabel('frequency')
ylabel('amplitude')

legend('AWGN snr=11dB','Clean Signal');

N=4096;
h =ones(4,4)/64;
sigma = 500;
gamma = 1;
alpha = 5;
ewx=wienerFilter(P,h,sigma,gamma,alpha);
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soundsc(ewx,Fs)
T=fft(P,4096);
Pyy=abs(T).^2;
dBs=10*log10(Pyy);
Ma=max(dBs)
%Measuring maxima of Noisy Signal
D=fft(ewx,4096);
Pyz=abs(D).^2;
dBS=10*log10(Pyz);
Maa=max(dBS) %Measuring maxima of reconstructed signal
figure(2),plot(ewx,'r'),
xlabel('frequency')
ylabel('amplitude')
legend('Reconstructed Signal');

Simulation Results:

NOISY SIGNAL SNR=1dB

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Reconstructed Signal:

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Conclusion and Future Works:

On the basis of this work one can progress this technique to make other useful contribution in
signal noise removal using the basic filtering techniques. Digital filtering of audio signals can
be done to make audio signals as much free from noise as possible keeping in mind that the
quality of the audio signal must be kept as pure as possible.

References:
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1. Li, Tan; Jean, Jiang; Improving digital signal processing course with realtime processing experiences for
electrical and computer engineering technology students, 2010 ASEE Annual
Conference and Exposition;
2010.
2. Barkana, Buket; A graduate level course: Audio Processing Laboratory,
2010 ASEE Annual Conference
and Exposition; 2010.
3. Adams, J.; Mossayebi, F.; Hands on experiments to instill a desire to learn
and appreciate digital signal
processing, 2004 Annual Conference and Exposition, 2004.
4. Ossman, Kathleen; MATLAB/Simulink lab exercises designed for teaching
digital signal processing
applications, 2008 ASEE Annual Conference and Exposition; 2008.
5. Ossman, Kathleen; MATLAB exercises to explain discrete fourier
transforms, 2003 ASEE Annual
Conference and Exposition; 2003.
6. Pierre, John W.; Kubichek, Robert F.; Hamann, Jerry C.; Enhancing the
comprehension of signal
processing principles using audio exercises with MATLAB, 1999 ASEE
Annual Conference and
Exposition; 1999.
7. Atti, Venkatraman; Spanias, Andreas; Panayiotou, Constantinos; Song, Yu;
Teaching digital filter design
techniques used in high-fidelity audio applications, 2004 ASEE Annual
Conference and Exposition; 2004.
8. Piano Key Frequencies, Wikipedia,
http://en.wikipedia.org/wiki/Piano_key_frequencies
9. Musical Instrument Samples, Electronic Music Studios, University of Iowa,
http://theremin.music.uiowa.edu/MIS.html
10. www.owling.com

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