WOLDIA UNIVERSITY

FACULTY OF TECHNOLOGY

DEPARTMENT OF ELECTRICAL AND COMPUTER

ENGINEERING

(COMMUNICATION STREAM)

Title: Design and analysis of Acoustic Echo Cancellation

Group members ID

1. Tesfa Tadesse……………….. FOT (R) 1981/07

2. Ayenew Nuru……… ………. FOT (R) 0546/07
3. Yonas Azie………. ………… FOT (R) 1499/07
Advisor Name: Instructor ABERA H.
Semester project submitted to: Electrical & computer Engineering department
(Communication stream)
Submission date:
29th may 2018
Woldia Ethiopia
 DECLARATION
We hereby declare that we carried out the work" Designing and Analysis of Acoustic Echo
Cancellation" in the department of Electrical and computer engineering, University of woldia,
under the advisor of Mr. Abera H. (MSc in communication Engineering). All source of
knowledge used have been duly acknowledge.

Name Signature Date

Yonas Azie _______________ ___________

Ayenew Nuru _______________ ___________

Tesfa Tadesse _______________ ___________

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 APROVAL
It is approved that this project has been written in compliance with the formatting rules laid
down by electrical and computer engineering university of Woldia, Ethiopia.

……………………………………………………….

Advisor Signature Date

MR. ABERA H. __________ _________

Department head

MR. Ali Y. __________ _________

Examining Committee Members Signature Date

1. Mr. _______________________ ___________ __________

2. Mr. _______________________ __________ __________

3. Mr. _______________________ __________ __________

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 ACKNOWLEDGMENT
First of all our gratitude goes to the Almighty God for his positive will and help without which
the things we made here were value less. Secondly we, want to thank our teachers especially, our
advisor Mr. Abera H.( MSC in electrical and computer engineering) for his unlimited, truly
help and guide us from the very beginning of selecting the title and we also thank some other
lectures for helping us to do the project.

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 ABSTRACT
Acoustic echo cancellation is important for audio teleconferencing when simultaneous
communication (or full-duplex transmission) of speech is necessary. In acoustic echo
cancellation, a measured microphone signal contains two signals: - the near-end speech signal
the far-end echoed speech signal. The goal is to remove the far-end echoed speech signal from
the microphone signal so that only the near-end speech signal is transmitted back to the far-end
listener. Acoustic echo cancellation is a common occurrence in today’s telecommunication
systems. The signal interference caused by acoustic echo is distracting to users and
causes a reduction in the quality of the communication. Acoustic echo path are more non-
stationary due to opening and closing of a door or movement of people inside the room, so the
echo canceler must accurately estimate the echo path characteristic and rapidly adapt to its
variation. This involves the selection of an adaptive filter and an algorithm for the adaptation.
Adaptive filtering constitutes one of the core technologies in digital signal processing and
finds numerous application areas in science as well as in industry. Adaptive filtering
techniques are used in a wide range of applications, including echo cancellation, adaptive
equalization, adaptive noise cancellation, and adaptive beam forming.

This paper will focus on the occurrence of acoustic echo in telecommunication systems and
adaptive filters are dynamic filters which iteratively alter their characteristics in order to achieve
an optimal desired output and implementing the overall system of acoustic echo cancellation
system using mainly Least Mean Square (LMS) and also little about Normalized Least Mean
Square (NLMS) algorithms for adaptive filter. It is the result of a project in the course adaptive
signal processing at Woldia University.

Finally With the world shrinking into a global village because of superior communications,
telephones, both conventional and hands-free sets, occupy a prominent position in solving
people’s communication needs, so The Echo cancellation algorithm presented in this paper
successfully attempted to find a software solution for the problem of echoes in the
telecommunications environment. The algorithm was capable of running in any PC with
MATLAB software installed.

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 TABLE OF CONTENT

Contents
DECLARATION ........................................................................................................................................... i
APROVAL.................................................................................................................................................... ii
ACKNOWLEDGMENT.............................................................................................................................. iii
ABSTRACT................................................................................................................................................. iv
TABLE OF CONTENT ................................................................................................................................ v
LIST OF FIGURE....................................................................................................................................... vii
ACRONYMS ............................................................................................................................................. viii
CHAPTER ONE ........................................................................................................................................... 1
1. Introduction ............................................................................................................................................... 1
1.1 Background ......................................................................................................................................... 1
1.2 Statement of problem .......................................................................................................................... 3
1.3 Objective of the Study ........................................................................................................................ 3
1.3.1 General Objective ........................................................................................................................ 3
1.3.2 The specific objectives ................................................................................................................. 3
1.4 Scope of this project............................................................................................................................ 4
1.5 Literature review ................................................................................................................................. 4
1.6 Methodology ....................................................................................................................................... 6
1.7 Project organization ............................................................................................................................ 6
CHAPTER TWO .......................................................................................................................................... 8
2. Basic Echo Canceller ................................................................................................................................ 8
2.2 Acoustic Echo Cancellation in teleconference/Videoconference ....................................................... 8
2.3 Acoustic Echo Cancellation (AEC) .................................................................................................... 9
2.4 Adaptive filters.................................................................................................................................. 11
2.4.1 General block diagram of the adaptive filters ............................................................................ 11
2.4.2 Adaptive Algorithms .................................................................................................................. 12
2.5 LMS algorithm .................................................................................................................................. 13
2.6 Performance evaluation .................................................................................................................... 16
2.6.1 Convergence Test....................................................................................................................... 16

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CHAPTER THREE .................................................................................................................................... 17
3. SYSTEM DESIGN FOR ECHO CANCELLATION ............................................................................. 17
CHAPTER FOUR....................................................................................................................................... 18
4. SIMULATION AND RESULTS ............................................................................................................ 18
4.1 Discussion ......................................................................................................................................... 18
4.2 Results ............................................................................................................................................... 18
CHAPTER FIVE ........................................................................................................................................ 23
5. CONCLUSION AND RECOMMENDATIONS .................................................................................... 23
5.1 Conclusion ........................................................................................................................................ 23
5.2 Recommendations ............................................................................................................................. 23
REFERANCE ............................................................................................................................................. 24
APPENDEX................................................................................................................................................ 25

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 LIST OF FIGURE
Figure 1.1 principle of acoustic echo cancellation system [1] ...................................................................... 2

Figure2. 1 a Generic Adaptive Echo Canceller [3] ....................................................................................... 8

Figure2. 2 The basic model of adaptive filter [3] ....................................................................................... 10
Figure2. 3 block diagram of adaptive filter [2] ........................................................................................... 12
Figure2. 4 LMS Algorithm [3].................................................................................................................... 14
Figure2. 5 Generic LMS Algorithm [3] ...................................................................................................... 15

Figure3. 1 Flowchart for Echo Cancellation ............................................................................................... 17

Figure4. 1 plot of the impulse response ............................................................................................ 18

Figure4. 2 Plot of the Far-end Signal, ................................................................................................ 19
Figure4. 3 Plot of the Near-end Signal, .............................................................................................. 19
Figure4. 4 Plot of the microphone Signal ................................................................................................... 20
Figure4. 5 Microphone Signal is entered into Adaptive filter and output of AEC ..................................... 20
Figure4. 6 Normalized LMS method .......................................................................................................... 21
Figure4. 7 convergence performance of regular LMS ................................................................................ 21
Figure4. 8 convergence performance of regular NLMS ............................................................................. 22
Figure4. 9 comparing the LMS and NLMS convergence performance ...................................................... 22

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 ACRONYMS
AEC…………………………… acoustic echo cancellation

RIR………………………………room impulse response

AF……………………………….adaptive filters

LMS……………………………..least mean-square

FFT………………………………fast Fourier transform

FDAF……………………………frequency-domain adaptive filter

DSP……………………………..digital signal processing

NLMS……………………………normalized least mean square

FE……………………………….far end

NE……………………………….near end

EMSE…………………………… estimate mean square error

ERLE……………………………...echo return loss enhancement

ANC………………………………active noise cancellation

FIR………………………………..finite impulse response

IIR…………………………………infinite impulse response

BIBO……………………………..bounded input bounded output

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i
 CHAPTER ONE

1. Introduction

1.1 Background
People have been using telephones as a way of distant voice communication for more than one
century until now. In a phone conversation, echo is the sound of one’s own voice being
played back to him after a delay. There are two types of echo presenting in typical
communication networks: hybrid (line) echo and acoustic echo. Acoustic echo cancellation and
line echo cancellation both address similar problems, and are often based on the same
technology. However, a line echo canceller generally cannot replace an acoustic echo canceller,
because acoustic echo cancellation is a more difficult problem. Hybrid echo is the outcome of
impedance mismatches along a telephone line, and is relatively easy to identify and cancel.
Acoustic echo is due to the leakage from loudspeaker to microphone in a hands-free telephone,
and is much more difficult to tackle. With acoustic echo cancellation, the echo path is very
complex (dozens or hundreds of reflections), lasts 100-200 ms, and can vary continuously during
a conversation as people move around the room. Acoustic echo cancellers are therefore much
more complicated devices. [1].

The solution to these problems is the elimination of the echo with an echo suppression or echo
cancellation algorithm. The echo suppressor offers a simple but effective method to counter the
echo problem. However, the echo suppressor possesses a main disadvantage since it supports
only half-duplex communication. Half-duplex communication permits only one speaker to talk at
a time. This drawback led to the invention of echo cancellers. An important aspect of echo
cancellers is that full-duplex communication can be maintained, which allows both speakers to
talk at the same time.

This has led to intensive research into the area of echo cancellation, with the aim of providing
solutions that can significantly reduce or even remove acoustic echo [1-3]. Good acoustic echo
cancellation system can greatly enhance the audio quality of a communication system by
allowing conversation to progress more smoothly and naturally, keeping the participants

1
more comfortable, and preventing listener fatigue. However, a poorly designed acoustic echo
cancellation will not provide these benefits and can even degrade audio quality significantly.
This paper describes the design and simulation of an acoustic echo cancellation (AEC)
approach, which is based on the least mean square(LMS) and somewhat normalized least
mean square (NLMS) algorithm. According to the simulation, the proposed acoustic echo
cancellation approach works well in the modest linear noisy environment.

The way how acoustic echo canceller works is shown in Figure 1.1. In general, the near
end microphone signal consists of signals from near end microphone, near end,
background noise and echo of the far-end audio signal originating from the loudspeaker
[1]. The last signal is undesired. The objective of the echo canceller is thus to form a replica of
the acoustic echo signal picked up by the terminal microphone by inserting an adaptive
filter h(n) parallel to the signal path through loudspeaker, room, and microphone [4].

The objective of the adaptive filter is to provide a response as equal as possible to that of the
acoustic signal path. The far end speech signal is then fed through the adaptive filter
to generate the signal y(n) to resemble the echo part of the microphone signal. The
echo is then subtracted from the microphone signal to get the desire speech signal

Figure 1.1 principle of acoustic echo cancellation system [1]

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1.2 Statement of problem
Our university is making relation with other universities with in country and outside the country
but still now we cannot get on line learning and training, so we are assume the limitation is there
is no good video and teleconferencing room and even it needs good echo cancellation. During
the meeting to address the information uniformly from one place by the president or selected
person to the whole student attend in meeting hall is still impossible in our university.

The noise i.e. echo is now available everywhere which can harm or change the original signal
and a reduction in the quality of the communication. Noise can’t be removed totally but can be
reduced and controlled by using different mechanisms Acoustic echo cancellation traditionally
involves passive methods such as enclosures, barriers and silencers to attenuate noise. These
techniques use either the concept of impendence change or the energy loss due to sound
absorbing materials. These methods are however not effective for low frequency noise. Many
factors influence the design of an AEC system, such as computational complexity, memory
consumption. A technique to overcome this problem is Acoustic echo Cancellation (AEC), so we
design of Adaptive Acoustic Echo Canceller Using LMS Algorithm in MATLAB software,
which is sound field modification by elect acoustic means.

1.3 Objective of the Study

1.3.1 General Objective

The general objective is to design and implementation of acoustic echo cancellation using
MATLAB software.

1.3.2 The specific objectives

 To understand how echo cancelling is achieved.

 To develop and implement the active echo cancellation in accordance to the established
design parameters.
 Design of Adaptive Acoustic Echo Canceller Using LMS Algorithm.
 To cancel or reduce the echo signal as much as possible.

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1.4 Scope of this project
Scope of this project is about echo cancelers have found wide use in full-duplex data
transmission over two-wire circuits and are now being applied to control echo and howl in
electronic conferencing applications. We simulated using adaptive algorithm to reduce echo in a
communication system by MATLAB software.

Implement an existing code to reduce Acoustic Echo by using different Algorithms. The
variations in the echo path models were small. They do not vary as much as required during the
process. The improvement on echo canceller with very low complexity algorithms to work well
even with big variations in echo path models and in noisy environment will be the further scope.

1.5 Literature review

The aim of this work is to review the most recent acoustic echo cancellation techniques
and their applicability for current hands free applications. Therefore, this paper presents
Acoustic echo cancellation (AEC) systems challenges and comparison between these techniques
is also presented.

The history of echo cancellation begins on 10 July 1962.On that date, Bell Laboratories ushered
in the era of satellite communications with the launching of Telstar, the first telecommunications
satellite. Echo cancellation was developed in the early 1960s by AT&T Bell Labs and
later by COMSAT Tele-Systems. The adaptive echo canceller tries to overcome the
deficiencies of the classical methods [1] (Sondhi, Mar. 1967.[2])].

[1]Telephony via Telstar (mentioned earlier), which was a low-flying satellite at about 200 mi
above the surface of the Earth, would not have introduced any major problems beyond those of
terrestrial telephony. However, even in the early 1960s, it was anticipated that telephony via
satellites would eventually use so-called geostationary satellites. These are satellites in the
earth’s equatorial plane that take exactly 24 h to complete an orbit, so they stay at a fixed
position relative to the earth. That corresponds to about 500 ms of propagation time for a radio
wave. With additional delays at the end point, the total round trip delay of the echo can be as
high as 600 ms. Experiments showed that a delay of this length in the absence of echoes can be
acceptable for telephony. However, such a long delay changes the dynamics of a conversation to
the extent that echo suppressors no longer provide a satisfactory solution to the echo problem.

4
Bursts of unsuppressed echo and clipping of soft segments of the near-end speech become much
too frequent to be acceptable.

Improvements in the design of echo suppressors were not able to deal with this problem. A new
technique was required that did not interrupt the echo return path, and that technique was echo
cancellation as opposed to echo suppression.

The idea of adaptive echo cancellation, first proposed by John L. Kelly of Bell Labs. John Kelly
unfortunately died of a sudden heart attack so the task of writing the first full paper summarizing
this work fell on Man Mohan Sondhi[1] (Sondhi, Mar. 1967.[2])

M.M Sondhi clearly states the history of echo cancellation that is important used as a
background for this project and also idea of adaptive echo cancellation by using different
algorithm was addressed. However he did not evaluate the effective working of the algorithm
and compare the convergence performance of regular LMS algorithm to the normalized variant
also did not addressed by him.

Therefore in order to evaluate the effective working of the algorithm, some basic tests were
conducted that is convergence test and Compare the convergence performance of regular LMS
algorithm to the normalized variant in this project were added.

Lin Bai et al. adaptive algorithm for ANC to filter a noisy speech signal is proposed in this
paper. The new algorithm further relaxes the constraint in the CS-LMS algorithm. The new
algorithm attempts to minimize the estimation error and the estimation is obtained using the
concept of Taylor’s expansion. CS-LMS can be seen as a special case of the new
algorithm. Analysis results show that the new algorithm can obtain better EMSE performance
than NLMS and the new algorithm can get performance improvement. [2]

Minimization the estimation error and performance increasing is achieving in a good manner
were considered as advantage. However it leads additional cost due to computational complexity
that is used Taylor’s explanation. Therefore in this project minimize the computational
complexity and specifying the project in echo cancellation not consider the whole noise that is
modified from Lin Bai et al.

5
From the literature review, the input that has been collected is useful for better understanding of
this project to overcome the problem caused by echo cancellation is used. Now that the need for
an acoustic echo canceller is recognized, this project gives the best AEC solution for our
university by MATLAB software. An overwhelming breadth of the literature, curving
different techniques for Acoustic echo cancellation the project report’s in the area of
Acoustic echo cancellation for woldia university.

1.6 Methodology
We had to start from the research phase at the beginning and need to gain knowledge on all the
echo cancellation and different type algorisms components that we had intended to use for our
project.

Generally, the methods and steps involved throughout the study are as following;
i. Literature review: We are gathering different information and components by referring
different books and searching web site regarding with our project.
ii. System analysis and design approach: By using LMS Algorism we will design acoustic
echo cancellation and analyze how to remove delayed (echo) part from original sound.

iii. Simulation: Simulate in MATLAB software

iv. Hardware tools: Desktop computer (laptop), Storage devices: hard disk, flash disc,
displaying devices like printer.

1.7 Project organization

This paper analyzed the echo cancellation by developing MAT LAB code. The project is
preceded by the use of MAT LAB software. Generally this project includes five chapters, each
describing different parts but in general related.

The first chapter introduces the background, statement of problem; objectives of the project
including the scope and limitation of the project, literature review and methodology were
explained.

The second chapter explores the basics of acoustic echo cancellation, the need and advantage of
the echo cancellation. It also explains about adaptive filters application and different algorithms.

6
The third chapter is about the system design and flowchart for echo cancellation, the analysis of
LMS algorithm and its block diagram are explained.

The fourth part is the result and discussion of the project, in this part the MATLAB Code is
developed and run then the plots are put with explanation of their characteristics. The output of
echo canceller and comparison of the convergence performance is shown in this part by
simulating in MAT LAB software.

Finally the last part is about conclusion and recommendation based on result as this is one parts
of the projects.

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 CHAPTER TWO

2. Basic Echo Canceller

[3]A basic echo canceller used to remove echo in telecommunication networks is presented in
Figure 2.1. This problem can be solved by using an adaptive filter that gradually matches its
estimated impulse response, to that of the impulse response of the actual echo path, h. The echo
path is highly variable and can even depend on such things as the movement of people in the
room as well as other things. These variations are accounted for by the adaptive control loop,
which is built into the canceller. The estimated echo , desired response
, error signal is ) is given by

In the ideal case, this represents the case when the adaptive echo canceller is
perfect.

Figure2. 1 a Generic Adaptive Echo Canceller [3]

2.2 Acoustic Echo Cancellation in teleconference/Videoconference

[3]When the telephone connection is between hands-free telephones or between two conference
rooms, then an acoustic echo problem emerges that is due to the reflection of the loudspeaker’s

8
sound waves from the boundary surfaces and other objects back to the microphone. This acoustic
echo can be removed using an adaptive filter as illustrated in Figure 2.1 Echo is the delayed and
distorted version of the original speech which is reflected back to the source. If reflected wave
arrives after a very short time of direct sound, it is considered as a spectral distortion or
reverberation. However, when the reflected wave arrives a few tens of milliseconds after the
direct sound, it is heard as a distinct echo. In data communication, the echo can lead to a big data
transmission error. In application like hands-free telecommunications, this echo, in
multichannel conditions, can make conversations impossible. Thus the echo has been a big
problem in communications networks.

2.3 Acoustic Echo Cancellation (AEC)

Acoustic echo occurs when an audio signal is reverberated in an enclosed environment such as
conference rooms. The echo signal is the combination of original signal plus attenuated and time
delayed images of the original signal. In this paper the echo path is generated using image model
of the Room Impulse Response (RIR).

[4]Adaptive filters are efficient filters that iteratively alter their filter coefficients in order to
achieve an optimal output. Adaptive echo cancellation is achieved by synthesizing the effect of
echo path on voice or data and subtracting it from the echo path output. The error function
which is the difference between the desired Signal and the filtered output is minimized
algorithmically by the adaptive filter by altering the coefficients. For the cancellation of echo
the adaptive filter is used in the feedback path. The role of the adaptive filter is to minimize
the error between the desired signal (i.e., the signal reverberated within the acoustic
environment) and filter output. The error signal is used to steer the filter coefficients to converge
fatly to optimum value which is depend on the input signals back into the adaptive filter and its
coefficients are converged according to an update equation to minimize this error function or
cost function. In case of AEC, the optimum value of the output of the adaptive filter is equal to
the echo signal. While the adaptive filter output is equal to the desired signal the error signal
goes to zero. In this particular situation as we want the echo signal is completely cancelled and
the FE user would not receive the original speech returned back to them.

9
 Figure2. 2 The basic model of adaptive filter [3]

The notations are used in the figure above are:

Far-end signal:

Near-end signal:

The true echo path (room impulse response):

Echo signal

Microphone signal:

Estimate echo path:

Estimate echo signal

Error signal: –

The echo path h of the room normally variable depends on the room structure and the moving
object inside. The estimated echo ŷ(n) is calculated from the reference input signal x(n)
and the adaptive filter ĥ(n). The near-end signal v(n) and background noise are added
into echo signal y(n) to create the desired signal d(n) ,

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The signal x (n) and y (n) are correlated. Then, it gets the error signal as,

The adaptive filter works to minimize the echo – to be zero to obtain only near-
end signal in the perfect case.

2.4 Adaptive filters

[4]The adaptive filters can be implemented in a number of different structures or realizations.
The choice of the structure can influence the computational complexity of the process and also
the necessary number of iterations to achieve a desired performance level. Basically, there
are two major classes of adaptive digital filter realizations, distinguished by the form of
the impulse response, namely the FIR filter and the Infinite duration Impulse Response (IIR)
filters. FIR filters are usually implemented with non-recursive structures, whereas IIR filters
utilize recursive realizations. FIR filter is preferred for AEC applications for its stability
during adaptation[1]. IIR can normally achieve similar performance as FIR, with smaller
amount of coefficients and less computation. However, as the complexity of the filter grows, the
order of the IIR filter increases a lot and the computational advantage is less dominant. Also, IIR
suffers from the instability problem.

2.4.1 General block diagram of the adaptive filters

The Wiener filter consists of a digital input signal an desired output and a linear filter
containing the answer to the impulse the input is used to obtain the output ) [4].
The difference between the output of Wiener filter and the desired output is the
estimation error e(k).The importance of Weiner filtration is given by the answer of the
impulse ), which gives a small estimation error. So, the main purpose is to minimize the
square value of

Here w represents the coefficients of the FIR filter tap weight vector, x(n) is the input vector
sample, is a delay of one sample, y(n) is the adaptive filter output, d(n) is the desired echoed
signal and ) is the estimation of the error signal at time n. The aim of an adaptive filter is to
calculate the difference between the desired signal and the adaptive filter output, e(n). The error

11
signal is fed back into the adaptive filter and its coefficients are changed algorithmically in order
to minimize a function of this difference, which is known as the cost function.

Figure2. 3 block diagram of adaptive filter [2]

2.4.2 Adaptive Algorithms

The choice of algorithm is determined by the performances like rate of convergence, miss-
adjustment, tracking, robustness, computational complexity, structure. Adaptive algorithms are
broadly classified as sample-by-sample adaptive and block adaptive algorithms [4].

In the sample-by-sample adaptive algorithms the adaptation can take place both in time
domain as well as in frequency domain. Therefore, sample-by sample adaptive algorithm will
further divided into two classes of algorithms. One class includes filter that are updated in time
domain sample-by-sample, called time domain sample-by sample adaptive algorithm.
Algorithms belonging to this category are Least Mean Square (LMS) and Recursive Least mean
Square (RLS) which are popular due to number of advantages [1]. Other class includes filters
that are updated in frequency domain and are known as Frequency Domain Adaptive Filter
(FDAF) [3]. FDAF is designed to achieve fast convergence rate and low computational cost
using of FFT reduces the computational complexity of FDAF. The common algorithm belonging

12
to this category is frequency domain adaptive algorithm. Instead of linear convolution and
correlation that are performed in LMS and NLMS adaptive filter, circular operation is
performed in frequency domain on a block-by-block rule instead of sample by sample in LMS
and NLMS. The major drawbacks of FDAF is has long delay due to some restore
operation that is required to perform the circular operation [3].

Adaptive filter algorithms are widely applied in acoustic echo canceller (AEC) such as namely
Recursive Least Square (RLS) [2] filter and Least Mean Square (LMS) filter. The RLS is a
Recursive implementation of Wiener filter that is used for stationary processes [2]. A relatively
simpler algorithm, LMS, uses the gradient search method to search for the least square
error filter coefficients. The LMS algorithm provides good numerical stability and its hardware
requirements are low, therefore being the best choice on the available hardware platform. A
disadvantage of this algorithm is its weak convergence, but based on obtained results this didn’t
prove to become a major issue. On the other hand, the NLMS algorithm is one of the most
implemented adaptive algorithms in actual telecom/industrial applications. The echo cancellation
algorithm is recommended to be implemented in hardware in case of Internet conversations
(VoIP or voice chat) where high latencies and echo can induce perturbations.

2.5 LMS algorithm

[3]The least mean square, (LMS), is a search algorithm that is widely used in various
applications of adaptive filtering. The main features that attracted the use of the LMS
algorithm are low computational complexity, proof of convergence in stationary
environments and stable behavior when implemented with finite precision arithmetic. Figure
2.4 illustrates how such an algorithm works. A path that changes the signal x is called h. Transfer
function of this filter is not known in the beginning. The task of the LMS algorithm is to estimate
the transfer function of the filter. The result of the signal distortion is calculated by convolution
and is denoted by r. In this case r is the echo and h is the transfer function of the hybrid. The
near-end speech signal v is added to the echo. The adaptive algorithm tries to create a filter w.
The transfer function of the filter is an estimate of the transfer function for the hybrid. This
transfer function in turn is used for calculating an estimate of the echo. The echo estimate is
denoted by rˆ [3].

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 Figure2. 4 LMS Algorithm [3]

The signals are added so that the output signal from the algorithm is

Where denotes the error signal. The error signal and the input signal x are used for estimation
of the filter coefficient vector w. One of the main problems associated with choosing the filter
weight is that the path h is not stationary. Therefore, the filter weights must be updated
frequently so that the adjustment to the variations can be performed.

The filter is a FIR filter with the form

A perfect FIR filter is linear, time-invariant and stable in a BIBO sense.

However, in a real-time environment, linearity is never a possibility and the first criterion is not
fulfilled so the filter can never be perfect. Updating of the filter weights is realized in accordance
with

For 2….where represents an estimate of the gradient vector and µ is the

convergence factor or step size.

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 Figure2. 5 Generic LMS Algorithm [3]

Figure 3.5 shows that

– –

Where ) is a vector containing the filter weights [ and ) represents

the . L is the length of the adaptive filter.

Then the gradient estimate is given by

( )

The resulting gradient-based algorithm is known. It minimizes the mean of the squared error, as
the least-mean square (LMS) algorithm, whose updating equation is given by

15
2.6 Performance evaluation

2.6.1 Convergence Test

The first and paramount test of the algorithm was whether or not the algorithm converged. If the
filter coefficients used in the adaptive algorithm did not converge, the code would be useless.
Therefore, several tests were performed on the simulated data in order to verify the convergence
of the filter coefficients. These tests were conducted by varying the convergence factor, µn, and
examining the effect on the filter coefficients and the plot of the error signal ). Through
careful observation it was determined that produced faster convergence. And we can evaluate by
Echo Return Loss Enhancement (ERLE) this is measured in dB is defined as the ratio of the
instantaneous power of the signal, d(n), and the instantaneous power of the residual error signal,
e(n), immediately after cancellation.

For a good echo canceller circuit, an ERLE in the range of 30 dB – 40dB is considered to be
ideal. ERLE measures the amount of loss introduced by the adaptive filter alone but in this
project evaluate only by Convergence Test.

16
 CHAPTER THREE

3. SYSTEM DESIGN FOR ECHO CANCELLATION

The echo cancellation system design flowchart is shown in following figure 5. This flow
chart is designed for acoustic echo cancellation system using LMS algorithm. This program must
create echo signal and desired signal d (n). From overall flow chart, the process must read far-
end signal, x (n) and room impulse response, h. Then, the program created echo signal from
near-end speech signal, v (n) and read near end signal. The program must create desired signal, d
(n) to compare the result from the output results. The echo canceller run using LMS and NLMS
algorithm to compute error signal. Then, the echo cancellation process gets residual echo by
subtracting estimate echo from the desired signal. Finally, the nonlinear processor removed the
remaining echo from adaptive filter output.

Figure3. 1 Flowchart for Echo Cancellation

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 CHAPTER FOUR

4. SIMULATION AND RESULTS

4.1 Discussion
The previous chapters provide a detailed sketch of an AEC and the flowchart for the software
simulation. In this chapter the results of simulation of the AEC algorithm, which was performed
in MATLAB are discussed.

4.2 Results
First modeled the acoustic of the loudspeaker-to-microphone signal path where the speakerphone
is located. Use a long finite impulse response filter to describe the characteristics of the room.
The plot of impulse response is presented in Figure 4.1

Figure4. 1 plot of the impulse response

This section presents a graphical representation of the results obtained by simulating the
algorithm in MATLAB. The plot of the far-end signal ) is presented in Figure 4.2.

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 Figure4. 2 Plot of the Far-end Signal,

The plot of the near-end signal ) is presented in Figure 4.3.

Figure4. 3 Plot of the Near-end Signal,

The far-end signal was delayed and scaled in order to produce the echo signal. The signal at the
microphone contains both the near-end speech and the far-end speech that has been echoed
throughout the room. The goal of the acoustic echo canceler is to cancel out the far-end speech,
such that only the near-end speech is transmitted back to the far-end listener.

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 Figure4. 4 Plot of the microphone Signal

The desired signal, or Now this Microphone Signal is entered into Adaptive filter to get
desired Echo free signal which can be transmitted to the receiver end contain only near end
signal (Sender speech). After Filtering echo free Signals shown in figure 4.4. This algorithm is
very useful when the impulse response of the system to be identified is long. The FDAF uses a
fast convolution technique to compute the output signal and filter updates. This computation
executes quickly in MATLAB. Pick some initial parameters for the filter and see how well the
far-end speech is cancelled in the error.

Figure4. 5 Microphone Signal is entered into Adaptive filter and output of AEC

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By using Normalized LMS method in this paper can generate Echo free signal which can be
transmitted to the receiver end in figure 4.6

Figure4. 6 Normalized LMS method

In order to evaluate the effective working of the algorithm, some basic tests were conducted.
This section provides a brief account of these tests.

Compare the convergence performance of regular LMS algorithm to the normalized variant, we
see the normalized version adapts in far fewer iterations to a result almost as good as normalized
version.

Figure4. 7 convergence performance of regular LMS

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 Figure4. 8 convergence performance of regular NLMS

Figure4. 9 comparing the LMS and NLMS convergence performance

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 CHAPTER FIVE

5. CONCLUSION AND RECOMMENDATIONS

5.1 Conclusion
The Echo cancellation algorithm presented in this paper successfully attempted to find a
software solution for the problem of echoes. This paper is improves the overall system of
acoustic echo cancellation system using mainly LMS and also for comparison NLMS. The
performance of the LMS adaptive filtering algorithm is expressed by its simplicity to implement
and its stability when the step size parameter is selected appropriately, however the convergence
rate is poor. Therefore from the result concluded that the convergence rate of NLMS is better
than LMS but it introduce misadjustment.

5.2 Recommendations
There are many possibilities for further development in this work. Other techniques such as
infinite impulse response (IIR) or lattice filtering may prove to be more effective in an echo
cancellation application, but with more hardware complications and need more than one chip to
implement.

This project is not implementing in hardware or without utilizing any DSP hardware components
because the Speakerphone, Voice-Enabled Internet of Things (IoT) Equipment, Automatic
Speech Recognition (ASR) Preprocessing, very large scale integrated DSP processer needs high
power, high memory consumption and expensive even no available TMS320C6748 DSP low-
cost development kit (LCDK).

Therefore anyone having interest to support by budgets can implement this project in hardware
for the university or for any other in demand.

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 REFERANCE
[1] (Sondhi, Mar. 1967.[2])

[2] (Lin Bai, March, 2010.)

[3] (S. Raghavendran, October 2003)

[4] (H.HAYES, 1996)

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 APPENDEX
%on line sound importing or recourding
recObj = audiorecorder;
recordblocking(recObj, 15);
play(recObj);
y = getaudiodata(recObj);
plot(y);
play(recObj);
y = getaudiodata(recObj);
plot(y);

%code of AEC

M=4001;
fs=8000;
[B,A]=cheby2(4,20,[0.1, 0.7]);
Hd=dfilt.df2t([zeros(1,6) B]);
hFVT=fvtool(Hd);
set(hFVT, 'color' ,[1 1 1])
v=340;
H = filter(Hd,log(0.99*rand(1,M)+0.01).* ...
sign (randn(1,M)).*exp(-0.002*(1:M)));
H = H/norm(H)*4; % Room Impulse Response
plot(0:1/fs:0.5,H);
xlabel('Time [sec]' );
ylabel('Amplitude' );
title( 'Room Impulse Response' );
set(gcf, 'color' , [1 1 1])
figure(1)
hold on
load nearspeech
n = 1: length (v);
t=n/fs;
plot(t,v);
axis([0 33.5 -1 1]);
xlabel('Time [sec]' );
ylabel('Amplitude' );

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title('Near-end speech signal' );
set(gcf, 'color' , [1 1 1])
figure(2)
hold on
load farspeech
x =x( 1: length (x));
dhat = filter(H,1,x);
plot(t,dhat);
axis([0 33.5 -1 1]);xlabel('Time [sec]' );
ylabel('Amplitude' );
title('Far-End speech Signal' );
set(gcf, 'color' , [1 1 1])
figure(3)
hold on
d=dhat + v+0.001*randn(length(v),1);
plot(t,d);
axis([0 33.5 -1 1]);
xlabel('Time [sec]' );
ylabel('Amplitude' );
title('Microphon Signal' );
set(gcf, 'color' , [1 1 1])
figure(4)
hold on
mu=0.025;
W0 = zeros(1,2048);
del = 0.01;
lam = 0.98;
x = x(1:length(W0)*floor(length(x)/length(W0)));
d = d(1:length(W0)*floor(length(d)/length(W0)));
% Construct the Frequency-Domain Adaptive Filter
fdafilt = dsp.FrequencyDomainAdaptiveFilter('Length',32,'StepSize',mu);
[y,e] = fdafilt(x,d);
n = 1:length(e);
t = n/fs;
pos = get(gcf,'Position');
set(gcf,'Position',[pos(1), pos(2)-100,pos(3),(pos(4)+111)])
subplot(3,1,1);

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plot(t,v(n),'g');
([0 33.5 -1 1]);
xlabel('Time [sec]');
ylabel('Amplitude');
title('Near-End Speech Signal of MR.ABERA');
subplot(3,1,2);
plot(t,d(n),'b');
axis([0 33.5 -1 1]);
ylabel('Amplitude');
title('Microphone Signal Mr. Amex + Mr.Abera');
subplot(3,1,3);
plot(t,v(n),'r');
axis([0 33.5 -1 1]);
ylabel('Amplitude');axis
title('Output of Acoustic Echo Canceller');
set(gcf, 'color' , [1 1 1])
#%Normalized LMS method
FrameSize = 102; NIter = 14;
lmsfilt2 = dsp.LMSFilter('Length',11,'Method','Normalized LMS', ...
'StepSize',0.005);
firfilt2 = dsp.FIRFilter('Numerator', fir1(10,[.05, .075]));
sinewave = dsp.SineWave('Frequency',0.001, ...
'SampleRate',1,'SamplesPerFrame',FrameSize);
TS = dsp.TimeScope('TimeSpan',FrameSize*NIter,'TimeUnits','Seconds',...
'YLimits',[-3 3],'BufferLength',2*FrameSize*NIter, ...
'ShowLegend',true,'ChannelNames', ...
{'echo signal', 'Filtered signal'});
%%
% Pass the echo input signal into the LMS filter and view the filtered
% output in the time scope.
for k = 1:NIter
x = randn(FrameSize,1); % Input signal
d = firfilt2(x) + sinewave(); % echo + Signal
[y,e,w] = lmsfilt2(x,d);
TS([d,e]); % echo = channel 1; Filtered = channel 2

end

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#% convergence performance of regular NLMS

x = 0.1*randn(500,1);
[b,err,res] = fircband(12,[0 0.4 0.5 1], [1 1 0 0], [1 0.2],...
{'w' 'c'});
d = filter(b,1,x);
lms_normalized = dsp.LMSFilter(13,'StepSize',mu,...
'Method','Normalized LMS','WeightsOutputPort',true);
[~,e1,~] = lms_normalized(x,d);
plot([e1]);
title('NLMS Conversion Performance');
legend('NLMS Derived Filter Weights');
#% convergence performance of regular LMS

x = 0.1*randn(500,1);
[b,err,res] = fircband(12,[0 0.4 0.5 1], [1 1 0 0], [1 0.2],...
{'w' 'c'});
d = filter(b,1,x);
lms_normalized = dsp.LMSFilter(13,'StepSize',mu,...
'Method','LMS','WeightsOutputPort',true);
[~,e2,~] = lms_normalized(x,d);

plot([e2]);
title('LMS Conversion Performance');

legend('LMS Derived Filter Weights');

#% comparing the LMS and NLMS convergence performance

x = 0.1*randn(500,1);
[b,err,res] = fircband(12,[0 0.4 0.5 1], [1 1 0 0], [1 0.2],...
{'w' 'c'});
d = filter(b,1,x);
lms = dsp.LMSFilter(13,'StepSize',mu,'Method',...
'Normalized LMS','WeightsOutputPort',true);
lms_normalized = dsp.LMSFilter(13,'StepSize',mu,...

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 'Method','Normalized LMS','WeightsOutputPort',true);
lms_nonnormalized = dsp.LMSFilter(13,'StepSize',mu,...
'Method','LMS','WeightsOutputPort',true);
[~,e1,~] = lms_normalized(x,d);
[~,e2,~] = lms_nonnormalized(x,d);
plot([e1,e2]);
title('Comparing the LMS and NLMS Conversion Performance');
legend('NLMS Derived Filter Weights', ...
'LMS Derived Filter Weights','Location', 'NorthEast');

29