TGPCET/ECE/2015-16

CHAPTER I
Overview of Present Work
Introduction
1.1Need of Noise Cancellation
In this new age of global communications, wireless phones are regarded
as essential communications tools and have a direct impact on peoples dayto-day

personal

and

business

communications.

As

new

network

infrastructures are implemented and competition between wireless carriers

increases, digital wireless subscribers are becoming ever more critical of the
service and voice quality they receive from network providers. Subscriber
demand for enhanced voice quality over wireless networks has driven a new
and key technology termed echo cancellation, which can provide near wire
line voice quality across a wireless network.
Todays subscribers use speech quality as a standard for assessing the
overall quality of a network. Regardless of whether or not the subscribers
opinion is subjective, it is the key to maintaining subscriber loyalty. For this
reason, the effective removal of hybrid and acoustic echoes, which are
inherent within the telecommunications network infrastructure, is the key to
maintaining and improving the perceived voice quality of a call. Ultimately,
the search for improved voice quality has led to intensive research into the
area of echo cancellation. Such research is conducted with the aim of
providing solutions that can reduce background noise and remove hybrid
and acoustic echoes before any transcoder processing occurs. By employing
echo cancellation technology, the quality of speech can be improved
significantly. This chapter discusses the overall echo problem. A definition of
echo precedes the discussion of the fundamentals of echo cancellation and
the voice quality challenges encountered in todays networks.
1.2 Basics of Echo
Echo is a phenomenon where a delayed and distorted version of an
original sound or electrical signal is reflected back to the source. With rare

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exceptions, conversations take place in the presence of echoes. Echoes of

our speech are heard as they are reflected from the floor, walls and other
neighbouring objects. If a reflected wave arrives after a very short time of
direct sound, it is considered as a spectral distortion or reverberation.
However, when the leading edge of the reflected wave arrives a few tens of
milliseconds after the direct sound, it is heard as a distinct echo [1].
Since

the

advent

of

telephony

echoes

have

been

problem

in

communication networks. In particular, echoes can be generated electrically

due to impedance mismatches at various points along the transmission
medium. The most important factor in echoes is called end-to-end delay,
which is also known as latency. Latency is the time between the generation
of the sound at one end of the call and its reception at the other end. Round
trip delay, which is the time taken to reflect an echo, is approximately twice
the end-to-end delay.

Echoes become annoying when the round trip delay

exceeds 30 ms. Such an echo is typically heard as a hollow sound. Echoes

must be loud enough to be heard. Those less than thirty (30) decibels (dB)
are unlikely to be noticed. However, when round trip delay exceeds 30 ms
and echo strength exceeds 30 dB, echoes become steadily more disruptive.
However, not all echoes reduce voice quality. In order for telephone
conversations to sound natural, callers must be able to hear themselves
speaking. For this reason, a short instantaneous echo, termed side tone, is
deliberately inserted. The side tone is coupled with the callers speech from
the telephone mouthpiece to the earpiece so that the line sounds connected.
1.3 Types of Echo
In telecommunications networks there are two types of echo. One
source for an echo is electrical and the other echo source is acoustic [1].
The electrical echo is due to the impedance mismatch at the hybrids of a
Public Switched Telephony Network, (PSTN), exchange where the subscriber
two-wire lines are connected to four-wire lines. If a communication is simply
between two fixed telephones, then only the electrical echo occurs.
However, the development of hands-free teleconferencing systems gave rise
to another kind of echo known as an acoustic echo. The acoustic echo is due
to the coupling between the loudspeaker and microphone. These electrical
and acoustic echoes are discussed in greater detail in chapter 2.

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1.4 The Process of Echo Cancellation

An echo canceller is basically a device that detects and removes the
echo of the signal from the far end after it has echoed on the local ends
equipment. In the case of circuit switched long distance networks, echo
cancellers reside in the metropolitan Central Offices that connect to the long
distance network. These echo cancellers remove electrical echoes made
noticeable by delay in the long distance network.
An echo canceller consists of three main functional components:
Adaptive filter
Doubletalk detector
Non-linear processor
1.4.1 Adaptive Filter
The adaptive filter is made up of an echo estimator and a subtractor.
The echo estimator monitors the received path and dynamically builds a
mathematical model of the line that creates the returning echo. The model
of the line is convolved with the voice stream on the receive path. This
yields an estimate of the echo, which is applied to the subtractor. The
subtractor eliminates the linear part of the echo from the line in the send
path. The echo canceller is said to converge on the echo as an estimate of
the line is built through the adaptive filter.
1.4.2 Doubletalk Detector
A doubletalk detector is used with an echo canceller to sense when
far-end speech is corrupted by near-end speech. The role of this important
function is to freeze adaptation of the model filter when near-end speech is
present. This action prevents divergence of the adaptive algorithm.
1.4.3 Nonlinear Processor
The non-linear processor evaluates the residual echo, which is nothing
but the amount of echo left over after the signal has passed through the
adaptive filter. The nonlinear processor removes all signals below a certain
threshold and replaces them with simulated background noise which sounds
like the original background noise without the echo.

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1.4.4 Preventing Clipping

Clipping occurs during a telephone conversation when part of the
speech is erroneously removed. Clipping results due to the lack of a precise
Non-Linear Processor, (NLP). Specifically, the NLP fails to start and stop at
the right time. Typically, an NLP does not respond rapidly enough to the
introduction of speech through the local end. It replaces parts of words with
background noise, which makes the conversation hard to follow. The same
can happen when the NLP confuses the fading of the voice level at the end
of a sentence with a residual echo.
1.5 Research Motivation and Thesis Outline
Since echo cancellation is a very demanding process, real-time
implementation has only been possible through the use of custom very large
scale integration, (VLSI), processors or digital signal processors (DSP).
These processors are specially designed for signal processing tasks. They
provide parallel processing of commands and optimized pipeline structures.
However, since the computation power of regular home personal computers,
(PCs), has increased tremendously and powerful software has evolved, it is
now possible to perform real-time signal processing in the PC environment
as well. The advent of this growing capability was the motivation for this
research. The objective of the research was the implementation of a
software echo canceller running natively on a PC with the help of the
MATLAB software.
This thesis provides an overview of an improved echo cancellation
technique using a noise gate for the NLP. Chapter 1 discusses the definition
of echo, the necessity of echo cancellers in telecommunications network, the
basics of echo cancellation and the challenges of echo cancellation. Chapter
2 gives an overview of the types of echo and their sources. It also discusses,
in great detail, the echo phenomena in four major telecommunication
systems. The proposed echo cancellation algorithm is explained step bystep in chapter 3. Chapter 4 discusses the simulation of the proposed
algorithm, details of the simulation environment and the results obtained.
Finally Chapter 5 provides a summary and some ideas concerning further
work in this field.

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1.6 Reduction of Noise By Using Various Techniques

While there are dozens of different kinds of noise reduction
technique, some of them which is used up till now is as follows1.6.1 DOLBY and DBX noise reduction techniqueThe first widely used audio noise reduction was developed by RAY
DOBLY in 1966. Intended for professional use, Dolby Type A was an
encode/decode system in which the amplitude of frequencies in four bands
was

increased

during

recording

(encoding),

and

then

decreased

proportionately during playback (decoding). The Dolby B system was a

single band system designed for consumer products. In particular, when
recording quiet parts of an audio signal, the frequencies above 1 kHz would
be boosted. This had the effect of increasing the signal to noise ratio on tape
up to 10dB depending on the initial signal volume. The Dolby B system,
while not as effective as Dolby A, had the advantage of remaining listenable
on playback systems without a decoder.
Dbx was the competing analog noise reduction system developed by
dbx laboratories. There are two type of dbx, type I and type II. dbx Type I
and Type II are types of "companding noise reduction. It used a root-meansquared (RMS) encode/decode algorithm. However it could achieve up to 30
dB of noise reduction.
Uudecoded dbx playback also exhibited large amounts of dynamic
error, with audio levels going up and down constantly, making it a very
fatiguing experience.

Fig. 1.1: Dolby SR, Dolby A and dbx Type 1 noise reduction for all
Multitrack.

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1.6.2 Dynamic noise reduction techniqueDynamic Noise Reduction (DNR) is an audio noise reduction system,
introduced by National semiconductor to reduce noise levels on longdistance telephony. First sold in 1981, DNR is frequently confused with the
far more common Dolby noise reduction system. Because DNR is noncomplementary, meaning it does not require encoded source material, it can
be used to remove background noise from any audio signal, including
magnetic tape recordings and FM radio broadcasts, reducing noise by as
much as 10 dB.
The system never really got off the ground for three main reasons,
first and foremost while the system worked and especially the dynamic
expansion part[2].

Fig. 1.2: Dynamic noise reduction

1.6.3 MATLAB techniqueA MATLAB algorithm works in the time-frequency domain using some
linear or non-linear filters that have local characteristics and are often called
time-frequency filters. Noise can therefore be also removed by use of
spectral editing tools, which work in this time-frequency domain, allowing
local modifications without affecting nearby signal energy.
MATLAB is an interpreted language. The main disadvantage of
interpreted languages is execution speed. MATLAB is slow interpreter. There
are lack of library name conventions, lack of object orientated features and
in my opinion use to many memory[3].

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Fig. 1.3: MATLAB GUI for reduction of Noise in Human Voice

1.6.4 VHDL techniqueThis technique consists of an adaptive filter. An adaptive filter is a filter
that self-adjusts its transfer function according to an optimization algorithm
driven by an error signal. Because of the complexity of the optimization
algorithms, most adaptive filters are digital filters. By way of contrast, a
non-adaptive filter has a static transfer function.
Adaptive filters are required for some applications because some
parameters of the desired processing operation (for instance, the locations
of reflective surfaces in a reverberant space) are not known in advance. The
adaptive filter uses feedback in the form of an error signal to refine its
transfer function to match the changing parameters.
Generally speaking, the adaptive process involves the use of a cost
function, which is a criterion for optimum performance of the filter, to feed
an algorithm, which determines how to modify filter transfer function to
minimize the cost on the next iteration[4].

Fig. 1.4: Lab view for FPGA

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TABLE 1.1: Comparision Of Various Noise Reduction Techniques

Characteristics

Accuracy

Dolby and

DYNAMIC

dbx noise

noise

MATLAB

VHDL

reduction

reduction

technique

technique

technique

technique

Average

Average

Good

Better

Dolby and dbx.

Type I & Type
Root-meanMethod of reduction

squared (RMS)

of noise

encode/decode
algorithm.

II noise
reduction
systems, DNR
is a playbackonly signal
processing

Timefrequency
domain using

Adaptive filter for

some linear

reduction of noise.

or non-linear
filters

system .
These

It could achieve
Capacity

up to 30 dB of
noise reduction.

It can be
reduced noise
by as much as
10 dB

systems
actually add
noise to a
signal to
improve its

These system can

remove full error
from the Audio
signal

quality.

Application

It has

It is use in

It is used

revolutionized

ECG system

It is used in mobile

MP3,discs,

the way video

to read the

phones, digital

mini Discs etc.

surveillance

heart signal

cameras.

cameras deal.

in hospital.

CHAPTER II

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Literature Review
2.1 Ms. S. Savitha, Ms. S. Lakshmi, "Implementation of Efficient LMS
Adaptive Filter with Low-Adaptation Delay, IEEE Sponsored 2nd
International Conference On Electronics And Communication
Systems(ICECS 2015).
An efficient architecture for the implementation of a delayed least mean
square adaptive filter is presented here. For achieving lower adaptation
delay and area-delay-power efficient implementation, and propose a
strategy for optimized balanced pipelining across the time consuming
combinational blocks of the structure and proposed an efficient fixed-point
implementation scheme architecture, and derive the expression for steadystate error. The hardware design of LMS adaptive filter is proposed, to
overcome this steady state error by designing noiseless tap LMS Adaptive
filter.
2.2 Pramod Kumar Meher and Sang Yoon Park, Area-Delay-Power
Efficient Fixed-Point LMS Adaptive Filter With Low Adaptation-Delay,
IEEE Transactions on Very Large Scale Integration (VLSI) Systems,
Vol. 22, no. 2, February 2014.
This paper presents an efficient architecture for the implementation of a
delayed least mean square adaptive filter. For achieving lower adaptationdelay and area-delay-power efficient implementation, here used a novel
partial product generator and propose a strategy for optimized balanced
pipelining across the time-consuming combinational blocks of the structure.
From synthesis results, we find that the proposed design offers nearly 17%
less area-delay product (ADP) and nearly 14% less energy-delay product
(EDP) than the best of the existing systolic structures, on average, for filter
lengths

8,

16,

and

32.

We propose

an

efficient

fixed-point

implementation scheme of the proposed architecture, and derive the

expression for steady-state error.

2.3 Amrita and Rajesh Mehra, "Embedded Design of an Efficient

Noise Canceller for Digital Receivers, International Journal of

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10

Engineering Science and Technology (IJEST), Vol. 3 No. 2 Feb 2011

ISSN : 0975-5462.
Adaptive Noise Cancelling is a technique within an electronic system to
remove the unwanted noise affecting the desired signal. Adaptive Noise
Canceller is been applied increasingly in modern communication as it has
the

advantages

of

easy

implementation

and

low

computational

complexity.ANC techniques can also be applied to high frequency signals,

multiplexed data coming from an array etc. but in all the high speed
applications, a software implementation of ANC usually doesnt meet the
required processing speed unless digital high speed DSP is used with
dedicated hardware implementation. Digital signal processors have a wide
variety of applications. Now days, its becoming increasingly important in
our daily life but it imposes the constraints on area, power, speed and cost.
So the design has to be carefully chosen. The most commonly used design
tools used for hardware implementation are Application Specific Integrated
Circuits (ASIC), Digital Signal Processors (DSP) and FPGAs.
2.4 Sayed. A. Hadei and M. lotfizad, "A Family of Adaptive Filter
Algorithms

in

Noise

Cancellation

for

Speech

Enhancement,

International Journal of Computer and Electrical Engineering, Vol. 2,

No. 2, April 2010 1793-8163.
In many application of noise cancellation, the changes in signal
characteristics could be quite fast. This requires the utilization of adaptive
algorithms, which converge rapidly. Least Mean Squares (LMS) adaptive
filters have been used in a wide range of signal processing application
because of its simplicity in computation and implementation. Recently
adaptive filtering was presented, have a nice trade off between complexity
and the convergence speed. This paper describes a new approach for noise
cancellation in speech enhancement using the two new adaptive filtering
algorithms named fast affine projection algorithm and fast Euclidean
direction search algorithms for attenuating noise in speech signals. The
simulation results demonstrate the good performance of the two new
algorithms in attenuating the noise.
2.5

Draghiciu Nicolae and Reiz Romulus, "Noise Cancelling in Audio

Signal with Adaptive Filter, ACTA ELECTROTECHNICA Volume 45,

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11

No.6, 2004.
In

this

paper

hard ware

implementation of an Adaptive Noise

Canceller (ANC) is presented. It has been synthesized within an FPGA,

using a modified version of the Least Mean Square (LMS) algorithm.
There are many possible implementations for an ANC, but the most widely
used employs (as the adaptive filter) a Finite Impulse Response (FIR) digital
filter, whose coefficients are iteratively updated using the LMS algorithm.
The author noticed that Adaptive Noise Canceller is then useful for
enhancing the S/N ratio of data collected from sensors working in noisy
environment, or dealing with potentially weak signals.
2.6 Bernard Widrow, John R. Glover, John M. McCool, Charles S.
Williams, Adaptive Noise Cancelling: Principles and Applications,
Proceedings of the IEEE, VOL. 63, NO. 12, December 1975.
This paper describes the concept of adaptive noise cancelling, an
alternative method of estimating signals corrupted by additive noise or
interference. The method uses a "primary "input containing the corrupted
signal and a "reference "input containing noise correlated in some
unknown way with the primary noise. The reference input is adaptively
filtered and subtracted from the primary input to obtain the signal
estimate. In this paper author discuss the various application of Adaptive
Noise Cancelling like cancelling 60-Hz Interference in Electrocardiography,
Cancelling

the

Donor

ECG

in

Heart-Transplant

Electrocardiography,

Cancelling Noise in speech Signals. Adaptive noise cancelling is a method of

optimal filtering that can be applied whenever a suitable reference input is
available, the principal advantages of the method are its adaptive capability,
its low output noise, and its low signal distortion.

CHAPTER III

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12

Formulation of Present Work

In Digital Signal Processing, a Filter is a device or process that
removes unwanted component from the signal. Most often, this means
removing some frequencies and not others in order to suppress interfering
signals and reduce background noise.
The term filter is often used to describe a device in the form of a piece
of physical hardware or software that is applied to a set of noisy data in
order to extract information about a prescribed quantity of interest. The
noise may arise from a variety of sources.

3.1 Filtering Methods

The usual method of estimating a signal corrupted by additive noise is to
pass it through a filter that tends to suppress the noise while leaving the signal
relatively unchanged i.e. direct filtering.

Fig. 3.1: Structure of Filter

Filters used for direct filtering can be either Fixed or Adaptive.
1. Fixed filters - The design of fixed filters requires a priori knowledge of both
the signal and the noise, i.e. if we know the signal and noise beforehand; we
can design a filter that passes frequencies contained in the signal and rejects
the frequency band occupied by the noise.
2. Adaptive filters - Adaptive filters, on the other hand, have the ability to
adjust their impulse response to filter out the correlated signal in the input.
They require little or no a priori knowledge of the signal and noise
characteristics. (If the signal is narrowband and noise broadband, which is
usually the case, or vice versa, no a priori information is needed; otherwise
they require a signal (desired response) that is correlated in some sense to the
signal to be estimated. Moreover adaptive filters have the capability of
adaptively tracking the signal under non-stationary conditions.

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13

3.2 Types of filters

3.2.1 FIR Filter
FIR stands for finite impulse response. FIR filter is a type of a signal
processing filter whose impulse response (or response to any finite length
input) is of finite duration, because it settles to zero in finite time. This is in
contrast to infinite impulse response (IIR) filters, which have internal
feedback and may continue to respond indefinitely (usually decaying). The
impulse response of an Nth-order discrete-time FIR filter (i.e. with a
Kronecker delta impulse input) lasts for N+ 1 sample, and then dies to zero.
Figure 3.2 shows the structure of a direct-form FIR filter, also known
as a tapped-delay-line or transversal filter, where z-1 denotes the unit delay
element and each wi(n) is a multiplicative gain within the system. In this
case, the parameters in W(n) correspond to the impulse response values of
the filter at time n. We can write the output signal y(n) as
L1

y (n)= w ( n ) x (ni)
i=0

= WT(n)X(n)
where X(n) = [x(n) x(n 1) _ _ _ x(n L + 1)]T denotes the input signal
vector and

denotes vector transpose. This system requires L multiplies and

L 1 adds to implement, and these computations are easily performed by a

processor or circuit so long as L is not too large and the sampling period for
the signals is not too short. It also requires a total of 2L memory locations
to store the L input signal samples and the L coefficient values, respectively.

Fig. 3.2: Structure of FIR Filter

3.2.2 IIR Filter
IIR stands for Infinite impulse response. IIR is a property of signal
processing systems. Systems with this property are known as IIR systems

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14

or, when dealing with filter systems, as IIR filters. IIR systems have an
impulse response function that is non-zero over an infinite length of time.
This is in contrast to finite impulse response (FIR) filters, which have fixedduration impulse responses. The simplest analog IIR filter is an RC filter
made up of a single resistor (R) feeding into a node shared with a single
capacitor (C). This filter has an exponential impulse response characterized
by an RC time constant.
IIR filters may be implemented as either analog or digital filters. In
digital IIR filters, the output feedback is immediately apparent in the
equations defining the output. Note that unlike FIR filters, in designing IIR
filters it is necessary to carefully consider the "time zero" case in which the
outputs of the filter have not yet been clearly defined.
Design of digital IIR filters is heavily dependent on that of their analog
counterparts

because

there

are

plenty

of

resources,

works

and

straightforward design methods concerning analog feedback filter design

while there are hardly any for digital IIR filters. As a result, usually, when a
digital IIR filter is going to be implemented, an analog filter (e.g. Chebyshev
filter, Butterworth filter, Elliptic filter) is first designed and then is converted
to a digital filter by applying discretization techniques such as Bilinear
transform or Impulse invariance.
The structure of a direct-form IIR filter is shown in Fig.3.3.

Fig. 3.3: Structure of IIR filter

3.2.3 IIR Filters VS FIR Filters-

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15

1) IIR filters are difficult to control and have no particular phase,

whereas FIR filters make a linear phase always possible.
2) IIR filters can be unstable, whereas FIR filters is always stable.
3) IIR filters can have limited cycles, but FIR has no limited cycles.
4) IIR filters are derived from analog, whereas FIR filters has no
analog history.
5) IIR filters make polyphase implementation possible, whereas FIR
filters can always be made casual
3.3. Adaptive Filters
Adaptive filters are digital filters with an impulse response, or transfer
function, that can be adjusted or changed over time to match desired
system characteristics. Unlike fixed filters, which have a fixed impulse
response, adaptive filters do not require complete a priori knowledge of the
statistics of the signals to be filtered. Adaptive filters require little or no a
priori knowledge and moreover, have the capability of adaptively tracking
the signal under non-stationary circumstances.
For an adaptive filter operating in a stationary environment, the errorperformance surface has a constant shape as well as orientation. When,
however, the adaptive filter operates in a non-stationary environment, the
bottom of the error-performance surface continually moves, while the
orientation and curvature of the surface may be changing too. Therefore,
when the inputs are non-stationary, the adaptive filter has the task of not
only seeking the bottom of the error performance surface, but also
continually tracking it.
3.3.1 Applications of Adaptive Filters
Perhaps the most important driving forces behind the developments in
adaptive filters throughout their history have been the wide range of
applications in which such systems can be used. Now discuss the forms of
these applications in terms of more-general problem classes that describe
the assumed relationship between d(n) and x(n). Our discussion illustrates
the key issues in selecting an adaptive filter for a particular task.

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16

3.3.1.1 System Identification

Considering Fig. 3.4, this shows the general problem of system
identification. In this diagram, the system enclosed by dashed lines is a
black box, meaning that the quantities inside are not observable from the
outside. Inside this box is (1) an unknown system which represents a
general input output relationship and (2) the signal (n), called the
observation noise signal because it corrupts the observations of the signal at
the output of the unknown system.

Figure 3.4: System identification.

Since the model typically chosen for the adaptive filter is a linear filter, the
practical goal of the adaptive filter is to determine the best linear model that
describes the input-output relationship of the unknown system. Such a
procedure makes the most sense when the unknown system is also a linear
model of the same structure as the adaptive filter, as it is possible that
y(n) = b^(n) for some set of adaptive filter parameters. For ease of
discussion, the unknown system and the adaptive filter both be FIR filters.
3.3.1.2 Channel Identification
In communication systems, useful information is transmitted from one
point to another across a medium such as an electrical wire, an optical fiber,
or a wireless radio link. Non idealities of the Transmission medium or
channel

distort

the

fidelity

of

the

transmitted

signals,

making

the

deciphering of the received information difficult. In cases where the effects

of the distortion can be modeled as a linear filter, the resulting smearing of
the transmitted symbols is known as inter-symbol interference (ISI). In such
cases, an adaptive filter can be used to model the effects of the channel ISI

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17

for purposes of deciphering the received information in an optimal manner.

In this problem scenario, the transmitter sends to the receiver a sample
sequence x(n) that is known to both the transmitter and receiver. The
receiver then attempts to model the received signal d(n) using an adaptive
filter whose input is
the known transmitted sequence x(n): After a suitable period of adaptation,
the parameters of the adaptive filter in W(n) are fixed and then used in a
procedure to decode future signals transmitted across the channel. Channel
identification is typically employed when the fidelity of the transmitted
channel is severely compromised or when simpler techniques for sequence
detection cannot be used.
3.3.1.3 Plant Identification
In many control tasks, knowledge of the transfer function of a linear
plant is required by the physical controller so that a suitable control signal
can be calculated and applied. In such cases, one can characterize the
transfer function of the plant by exciting it with a known signal x(n) and
then attempting to match the output of the plant d(n) with a linear adaptive
filter. After a suitable period of adaptation, the system has been adequately
modeled, and the resulting adaptive filter coefficients in W(n) can be used in
a control scheme to enable the overall closed-loop system to behave in the
desired manner.
In certain scenarios, continuous updates of the plant transfer function
estimate provided by W(n) are needed to allow the controller to function
properly.
3.3.1.4 Echo Cancellation for Long-Distance Transmission
In voice communication across telephone networks, the existence of
junction boxes called hybrids near either end of the network link hampers
the ability of the system to cleanly transmit voice signals. Each hybrid allows
voices that are transmitted via separate lines or channels across a longdistance network to be carried locally on a single telephone line, thus
lowering the wiring costs of the local network. However, when small
impedance mismatches between the long distance lines and the hybrid
junctions occur, these hybrids can reflect the transmitted signals back to

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18

their sources, and the long transmission times of the long-distance network
about 0:3 s for a trans-oceanic call via a satellite linkturn these
reflections into a noticeable echo that makes the understanding of
conversation difficult for both callers. The traditional solution to this problem
prior to the advent of the adaptive filtering solution was to introduce
significant loss into the long-distance network so that echoes would decay to
an acceptable level before they became perceptible to the callers.
Unfortunately, this solution also reduces the transmission quality of the
telephone link and makes the task of connecting long distance calls more
difficult.
An adaptive filter can be used to cancel the echoes caused by the
hybrids in this situation. Adaptive filters are employed at each of the two
hybrids within the network. The input x(n) to each adaptive filter is the
speech signal being received prior to the hybrid junction, and the desired
response signal
d(n) is the signal being sent out from the hybrid across the long-distance
connection.

The

adaptive filter

attempts

to

model the transmission

characteristics of the hybrid junction as well as any echoes that appear

across the long-distance portion of the network. When the system is
properly designed, the error signal e(n) consists almost totally of the local
talkers speech signal, which is then transmitted over the network. Such
systems were first proposed in the mid-1960s and are commonly used
today.
3.3.1.5 Acoustic Echo Cancellation
A related problem to echo cancellation for telephone transmission
systems

is

that

of

acoustic

echo

cancellation

for

conference-style

speakerphones. When using a speakerphone, a caller would like to turn up

the amplifier gains of both the microphone and the audio loudspeaker in
order to transmit and hear the voice signals more clearly. However, the
feedback path from the devices loudspeaker to its input microphone causes
a distinctive howling sound if these gains are too high. In this case, the
culprit is the rooms response to the voice signal being broadcast by the
speaker; in effect, the room acts as an extremely poor hybrid junction, in
analogy with the echo cancellation task discussed previously. A simple

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19

solution to this problem is to only allow one person to speak at a time, a

form of operation called half-duplex transmission. However, studies have
indicated that half-duplex transmission causes problems with normal
conversations, as people typically overlap their phrases with others when
conversing.
To maintain full-duplex transmission, an acoustic echo canceller is
employed in the speakerphone to model the acoustic transmission path from
the speaker to the microphone. The input signal x(n) to the acoustic echo
canceller is the signal being sent to the speaker, and the desired response
signal d(n) is measured at the microphone on the device. Adaptation of the
system occurs continually throughout a telephone call to model any physical
changes in the room acoustics. Such devices are readily available in the
marketplace today. In addition, similar technology can and is used to
remove the echo that occurs through the combined radio/room/telephone
transmission path when one places a call to a radio or television talk show.
Details of the acoustic echo cancellation problem can be found in.
3.3.1.6 Adaptive Noise Cancelling
When collecting measurements of certain signals or processes,
physical constraints often limit our ability to cleanly measure the quantities
of interest. Typically, a signal of interest is linearly mixed with other
extraneous noises in the measurement process, and these extraneous
noises introduce unacceptable errors in the measurements. However, if a
linearly related reference version of any one of the extraneous noises can be
cleanly sensed at some other physical location in the system, an adaptive
filter can be used to determine the relationship between the noise reference
x(n) and the component of this noise that is contained in the measured
signal d(n). After adaptively subtracting out this component, what remains
in e(n) is the signal of interest. If several extraneous noises corrupt the
measurement of interest, several adaptive filters can be used in parallel as
long as suitable noise reference signals are available within the system.
Adaptive noise cancelling has been used for several applications. One
of the first was a medical application that enabled the electroencephalogram
(EEG) of the fetal heartbeat of an unborn child to be cleanly extracted from
the much-stronger interfering EEG of the maternal heartbeat signal.

TGPCET/ECE/2015-16

20

Details of this application as well as several others are described in the

seminal paper by Widrow and his colleagues.
3.3.1.7 Channel Equalization
Channel equalization is an alternative to the technique of channel
identification described previously for the decoding of transmitted signals
across non ideal communication channels. In both cases, the transmitter
sends a sequence s(n) that is known to both the transmitter and receiver.
However, in equalization, the received signal is used as the input signal x(n)
to an adaptive filter, which adjusts its characteristics so that its output
closely matches a delayed version s(n 1) of the known transmitted signal.
After a suitable adaptation period, the coefficients of the system either are
fixed and used to decode future transmitted messages or are adapted using
a crude estimate of the desired response signal that is computed from y(n).
This latter mode of operation is known as decision-directed adaptation.
Channel equalization was one of the first applications of adaptive
filters.

Today, it remains as one of the most popular uses of an adaptive

filter. Practically every computer telephone modem transmitting at rates of

9600 baud (bits per second) or greater contains an adaptive equalizer.
Adaptive equalization is also useful for wireless communication systems.
Equalization is closely related to linear prediction.
3.4 Algorithms to Reduce Noise
Following are the various algorithms that help us to reduce noise in
speech signals.
1)
2)
3)
4)
5)
6)

LMS (Least Mean Square) Algorithm.

EM (Expectation Maximization).
NLMS (Normalized least mean squares) Algorithm.
VSLMS (Variable Step-size LMS) Algorithm.
VSNLMS (Variable step size normalized LMS) Algorithm.
RLS (Recursive least squares) Algorithm.

CHAPTER IV

TGPCET/ECE/2015-16

21

Design of Experimentation
4.1Concept of Adaptive Noise Cancelling Using LMS Algorithm:
Adaptive Process involves the automatic adjustment of the tap
weights of the filter in accordance with the estimation error.
Thus, the combination of two processes i.e. filtering and adaptive
process working together constitutes a feedback loop around the LMS
algorithm. The transversal filter, around which the LMS algorithm is built,
this component is responsible for performing the filtering process. The
other one is mechanism for performing the adaptive control process on the
tap weights of the transversal filter, hence the designation" adaptive weight
control mechanism". This LMS algorithm is developed in Xilinx ISE9.2i.

Fig. 4.1: Concept of Adaptive Noise Canceller

As the name implies, ANC is a technique used to remove an
unwanted noise from a received signal, the operation is controlled in
an adaptive way in order to obtain an improved signal-to noise ratio
(SNR). As shown in "Fig.4.1", an ANC is typically a dual- input, closedloop adaptive feedback system. The two inputs are: the primary input
signal d(n) (the desired signal corrupted by the noise) and the reference
signal x(n) (an interfering noise supposed to be uncorrelated with the
desired signal but correlated with the noise affecting the desired signal in
an unknown way).

TGPCET/ECE/2015-16

22

Fig. 4.1 shows the basic problem and the adaptive noise Canceling
solution to it. A signal s is transmitted over a channel to a sensor that also
receives a noise no uncorrelated with the signal. The combined signal
and noise s+no form the primary input to the canceller. A second sensor
receives a noise n1 uncorrelated with the signal but correlated in some
unknown way with the noise no. This sensor provides the reference
input to the canceller. The noise n1 is filtered to produce an output y that is
as close a replica as possible of no. This output is subtracted from the
primary input s + no to produce the system output z = s + no y
In the system shown in Fig. 4.1 the reference input is processed by
an adaptive filter. An adaptive filter differs from a fixed filter in that it
automatically adjusts its own impulse response. Thus with the proper
algorithm, the filter can operate under changing conditions and can
readjust itself continuously to minimize the error signal. The error signal
used in an adaptive process depends on the nature of the application.
In the system shown in Fig. 4.1 the reference input is processed by
an adaptive filter. An adaptive filter differs from a fixed filter in that it
automatically adjusts its own impulse response. Thus with the proper
algorithm, the filter can operate under changing conditions and can
readjust itself continuously to minimize the error signal. The error signal
used in an adaptive process depends on the nature of the application. In an
adaptive noise cancelling system, in other words, the system output serves
as the error signal for the adaptive process. It might seem that some prior
knowledge of the signal s or of the noises no and n1 would be necessary
before the filter could be designed, or before it could adapt, to produce
the noise cancelling signal y.

4.2 Implementation of LMS Algorithm

There are three steps involved in every iteration of LMS algorithm. The order
of these steps is:
i. The output of the FIR filter, y(n) is calculated

TGPCET/ECE/2015-16

23
N 1

w ( n ) x ( ni )
()=

i=0

=()()
ii. The value of the error estimation is calculated

()=()()
iii. The tap weights of the FIR vector are updated in preparation for the next
iteration

(+1)=()+2()()
4.3 Pipelining of FIR Filter Block
Pipelining means breaking a large task down into a sequence of stages
such that data moves through the stages like parts moving through a
factory assembly line. Each stage produces output used by the next stage,
and all stages operate concurrently, resulting in better performance than if
data had to be fully processed by the task before new data could begin
being processed.

Fig 4.2: Pipelined FIR Filter

Figure 4.3 shows a pipelined version of this design. We merely add
resisters between the first and second rows of adders. Since the purpose of
these registers is solely related to pipelining. They are known as pipeline

TGPCET/ECE/2015-16

24

registers, though their internal design is the same as any other register. The
computations between pipeline registers are known as stages. By inserting
those registers and thus creating a two-stage pipeline. the critical path has
been reduced from 4 ns down to only 2 ns. and so the fastest clock has a
period of at least 2 ns. Meaning a frequency of no more than 1/2 ns = 5(X)
MHz. In other words, just by inserting those pipeline registers. weve
doubled the performance of the design.

Fig 4.3 : Non Pipelined verses Pipeline Data paths

4.4 LMS Core Implementation

TGPCET/ECE/2015-16

25

Figure 4.4: Block diagram of the LMS core

The LMS core is divided into five blocks1. Control block
2. Delay Block
3. Multiply Accumulator (MAC) Block
4. Error Counting Block
5. Weight Update Block
4.4.1 Control block
The Control block arranges the timing of the whole system. It
produces four enable signals: en_x,en_d,en_coee, en_err, which enable the
Delay Block, the Weight Update Block and the Error Counting Block
separately. When read=1 all the enable signals get 1 that means it will read
input and produce output. When write=1 all the enable signals get 0 that
means it will not read any input just check output.
4.4.2 Delay block

TGPCET/ECE/2015-16

26

The Delay Block receives the reference signal x_in and the primary
input signal d_in under control of the enable signal en_x and en_d. And it
produces the M tap delay signal x_out. When enable signals get 1 output
follows the input otherwise it will produce delay singal.
4.4.3 MAC block
The Multiply Accumulator (MAC) Block multiply the M_tap reference
signal x_out with the M_tap weight w separately, and add them together,
then we get yn. In MAC block as when clk=1 and reset=1 then filter
output=0 and when clk=0 and reset=0 then filter output by multiplying
reference input and weight and add them separately with previous filter
output.
4.4.4 Error counting block
The Error Counting Block subtracts yn from dn and get the error signal
e_out, which is also the output of the whole system. And it produces signal
xemu as a feedback by multiplying e_out, x_out and the scaling factor u.
When enable signal en_err get 1 it will generate error signal eout and
feedback signal xemu.
4.4.5 Weight Update Block
The Weight Update Block updates the weight vector w(n) to w(n+1) that
will be used in the next iteration. When enable signal en_coee get 1 next
weight equal to current weight plus feedback signal otherwise next weight
equal to zero.

CHAPTER V

TGPCET/ECE/2015-16

27

Design of Experimental Setup

5.1

Hardware Description Language

There are two industry standard hardware description languages, VHDL and
Verilog.
Hardware structure can be modeled equally effectively in both VHDL
and Verilog. When modeling abstract hardware, the capability of VHDL can
sometimes only be achieved in Verilog when using the PLI. The choice of
which to use is not therefore based solely on technical capability but on:
1) Personal preferences
2) EDA tool availability
3) Commercial, business and marketing issues
The modeling constructs of VHDL and Verilog cover a slightly different
spectrum across the levels of behavioral abstraction; see Figure 3.1

Fig. 5.1: HDL modeling capability

5.1.1 Verilog
Verilog was the first modern hardware description language to be
invented.
It was created by Phil Moorby and PrabhuGoel during the winter of
1983/1984.
Verilog is a hardware description language (HDL) used to model
electronic systems. Verilog HDL, not to be confused with VHDL (a competing
language), is most commonly used in the design, verification, and
implementation of digital logic chips at the register-transfer level of

TGPCET/ECE/2015-16

28

abstraction. It is also used in the verification of analog and mixed-signal

circuits.
Hardware description languages such as Verilog differ from software
programming languages because they include ways of describing the
propagation of time and signal dependencies (sensitivity).
Verilog is based on language. The designers of Verilog wanted a
language with syntax similar to the C programming language, which was
already widely used in engineering software development. Verilog is casesensitive, has a basic preprocessor and equivalent control flow keywords
(if/else, for, while, case, etc.)[5].
5.1.2 VHDL
VHDL stands for VHSIC Hardware description language .VHSIC is itself
an abbreviation for very high speed integrated Circuit. This language was
first introduced in 1981 for the department of Defense (DOD) under the
VHSIC program. In 1983 IBM, Texas instruments and inter metrics started
to develop this language. In 1987 IEEE standardized the language. VHDL is
a hardware description language .It describes the behavior of an electronic
circuit or a system from which the physical circuit or system can then be
implemented. VHDL is intended for circuit synthesis as well as circuit
stimulation. VHDL is a programming language that allows one to model and
develop complex digital system in a dynamic environment by dataflow,
behavioral and structural style of modeling.
VHDL is a standard technology independent language and is therefore
portable and reversible. The 2 main immediate application of VHDL are in
field of FPGA (field programmable gate array) logic device and in the field of
ASIC (application specific integrated circuits).VHDL is based on Pascal and
ADA. It is an object-oriented language and therefore people familiar with C+
+ or PASCAL can grasp it easily. VHDL can wear many hats. It is being used
for documentation, verification, and synthesis of large digital designs. This is
actually one of the key features of VHDL, since the same VHDL code can
theoretically achieve all three of these goals, thus saving a lot of effort. In
addition to being used for each of these purposes, VHDL can be used to take
three different approaches to describing hardware. These three different
approaches are the structural, data flow, and behavioral methods of

TGPCET/ECE/2015-16

29

hardware description. Most of the time a mixture of the three methods is

employed. VHDL is strongly typed and is not case sensitive.
Not

only VHDL

methodology

band

is

a description language,

but

also

design

environment. Designers are building next-generation

design technologies on VHDL. The

emerging field of electronic design

automation will result in tools that allow developers to create designs

graphically at a high level of abstraction. The EDA tools will automatically
produce VHDL, which logic synthesis tools can, in turn, readily synthesize
into a gate- level design. Since VHDL allows a design to be independent of
design tools and technology or vendor of the end product, a circuit can be
designed and archived in VHDL, and later fabricated with the most advanced
technology.
VHDL is intended to provide a tool that can be used by the digital
systems community to distribute their designs in a standard format. Using
VHDL, they are able to talk to each other about their complex digital circuits
in a common language without difficulties of revealing technical details. It is
a standard

and unambiguous way of exchanging device and system models

so that engineers have a clear idea early in the design process where
components from separate contractors may need more work to function
together properly. It enables manufactures to document and achieve
electronic systems and components in a common format allowing various
parties to understand and participate in a systems development.
As a standard description of digital systems, VHDL is used as input
and output to various simulation, synthesis, and layout tools. The language
provides the ability to describe systems, network, and components at a very
high behavioral level as well as very low gate level. It also represents a topdown methodology and environment. Simulations can be carried out at any
level from a generally functional analysis to a very detailed gate-level
waveform analysis. Synthesis is currently carried out only at the register
level. A register transfer level (RTL) description of hardware consists of a
series of Boolean logic expressions. Each operator represents a gate or a
series of gates in a hardware realization. Once the VHDL design has been
decomposed down to the register level, a VHDL synthesizer can generate

TGPCET/ECE/2015-16

the

application

30

specified

integrated

circuit

(ASIC)

representation

or

schematic for the PC board layout.

Top-down design first describes a system at a very high level of
abstraction, like a specification. Designers simulate and debug the system at
this very high level before refining it into smaller components. The methods
describe each component

in the system. The design continues t6o be

refined and debugged until it is complete down to its lowest building block.
Mixed- level design occurs when some components are at a more
detailed level of description than
design

others. The advantage of the top-down

methodology is that engineers can discover and correct system

problems early in the design cycle. Engineers can concentrate on overall

design issues such as system requirements and timing. The tedious task of
gate-level design can be left to synthesis tool. The bottom line is reduced
cost and faster time to manufacturing [4].

5.2 XILINX ISE 9.2i

Xilinx, Inc. is an American technology company, primarily a supplier of
Programmable

Logic

Devices.

It

is

known

for

inventing

the

Field

Programmable Gate Array (FPGA) and as the first semiconductor company

with fables manufacturing model. Founded in Silicon Valley in 1984, the
company is headquartered in San Jose, California, with additional offices in
Dublin, Ireland, Singapore and Tokyo, Japan.
Xilinx designs, develops and markets programmable logic products
including integrated circuits (ICs), software design tools, predefined system
functions delivered as intellectual property (IP) cores, design services,
customer training, field engineering and technical support. Xilinx sells both
FPGAs and CPLDs programmable logic devices for electronic equipment
manufacturers

in

end

markets

such

as

communications,

industrial,

consumer, automotive and data processing.

Xilinx FPGAs can run a regular embedded OS (such as Linux or
vxWorks) and can implement processor peripherals in programmable logic.

TGPCET/ECE/2015-16

31

The Spartan series targets applications with a low-power footprint,

extreme cost sensitivity and high-volume; e.g. displays, set-top boxes,
wireless routers and other applications.
The Spartan-6 family is built on a 45-nanometer [nm], 9-metal layer,
dual-oxide process technology. The Spartan-6 was marketed in 2009 as a
low-cost solution for automotive, wireless communications, flat-panel
display and video surveillance applications.
During the "tech boom years," competitor Altera was the market
leader. Today, Xilinx customers represent just over half of the entire
programmable logic market, at 51%. Altera is Xilinx's strongest competitor
with 34% of the marks markets such as communications, industrial,
consumer, automotive and data processing.

TGPCET/ECE/2015-16

32

CHAPTER VI
Conduct of Experimentation
6.1 Adaptive Filter Framework
Since the characteristics of the acoustic noise source and the
environment are time varying, the frequency content, amplitude, phase, and
sound velocity of the undesired noise are non stationary. An ANC system
must therefore be adaptive in order to cope with these variations. Adaptive
filters adjust their coefficients to minimize an error signal and can be
realized as (transversal) finite impulse response (FIR), (recursive) infinite
impulse response (IIR), lattice, and transform-domain filters. The most
common form of adaptive filter is the transversal filter using the least meansquare (LMS) algorithm. Figure 6.1 shows a framework of adaptive filter.
Basically, there is an adjustable filter with input X and output Y. Our goal is
to minimize the difference betweend and Y, whered is the desired signal.
Once the difference is computed, the adaptive algorithm will adjust the filter
coefficients with the difference. There are many adaptive algorithms
available in literature, the most popular ones being LMS (least meansquare) and RLS (Recursive least squares) algorithms. In the interest of
computational time, we used the LMS.

Fig. 6.1: Adaptive Filter Framework

6.2 Challenges

TGPCET/ECE/2015-16

33

The left hand side of Figure 6.2 shows the system of ANC. The use of
adaptive filter for ANC application is complicated by the fact that the
summing junction represents acoustic superposition in the space from the
canceling loudspeaker to the error microphone, where the primary noise is
combined with the output of the adaptive filter. Hence, the model is
sensitive to phase mismatch. If phase mismatch occurs, even though the
inverse noise is produced, the noise you hear cannot be canceled out
thoroughly. Besides, the ANC is sensitive to uncorrelated noise. If the
outside micro receives the uncorrelated noise, the DSK will try to produce
the inverse of the uncorrelated noise, which will degrade the performance.
Hence there are three solutions to solve these difficulties. First, to
consider

the

delay

caused

by

DAC,

ADC,

anti-aliasing

filter,

and

reconstruction filter. Then modeled this effect with another filter S(z), into
the mathematical model as shown in the right plot of figure 6.2. Second, to
reduce the delay, use FPGA to implement DAC, ADC, and analog AAF/RF.
Finally, some protective mechanisms added to stabilize the ANC system.

Fig. 6.2: Challenges at summing junction and solutions of

system delay.

TGPCET/ECE/2015-16

34

CHAPTER VII
Formulation of Models
Description:
To compare the RLS and LMS algorithms we utilized and improved the
existing functional scheme from MATLAB, precisely the scheme of RLS and
LMS algorithms for adaptive noise cancellation, as is shown in the Figures
7.1 and 7.2. The Fig.7.2 stayed without changes, while the internal part of
schemes of RLS adaptive filters (Fig. 7.1, on the left) and of LMS adaptive
filters (Fig. 7.2, on the left) changed radically. The all scheme, as is shown
in the Fig. 7.1, is represented by one block, i.e. the block of MATLABfunction. Since every MATLAB-function has only one input, insert a
multiplexer, which all the input signals collects to the one vector.

Fig. 7.1: Block diagram of noise cancellation LMS

algorithm (on left) and RLS algorithm (on right)

In the first, the RLS1 function, this has one input u and one output y.
Then declare auxiliary matrixes P and vectors W. In the third line ascertain
size of input data. In the divide input data to appropriate vectors uin =
uk+1, yout = dk+1 and lambda. The constant lambda is auxiliary constant
(neglecting factor). Write to the auxiliary vector of weight Wo the weights
from former time step. Then we evaluate inaccuracy of prediction, the prior
remainder, according the relation: (dk+1uk+1 T wLS(k)). In set down
covariance matrix from previous time step into matrix Po. Next represents
the actualization of covariance matrix. Then actualize the weights of
adaptive filter. Actualized weights are inscribed to the output of function.
Take down the covariance matrix and weight vector for the use in the
following time step.

TGPCET/ECE/2015-16

35

Fig. 7.2: Detailed diagram of noise cancellation LMS

algorithm (on left) and RLS algorithm (on right)

From the simulation of RLS and LMS filters have found, that the
adaptation rate of both filters was nearly equal; these algorithms have
adapted approximately after 200 evaluation steps for the sinusoidal
harmonic input signal. The quality of disturbance and noise cancellation is
more evident after 1000 actualized steps, when can observe also some
differences in transmission characteristics. From observations implies, that
RLS adaptive filters give higher quality in the disturbance and noise
cancellation. From mentioned comparisons implies that LMS algorithms are
simpler in evaluation process, but they attain lower quality in the
cancellation of disturbing signals. Contrary, RLS algorithms achieve higher
quality in the disturbing signal cancellation, but they have large numerical
claims for RLS filter coefficient evaluation.

TGPCET/ECE/2015-16

36

CHAPTER VIII
Analysis of Result
8.1 Top Level RTL Schematic of ANC

Fig 8.1: Top Level RTL Schematic of ANC

8.2 Second Level RTL Schematic of ANC

Fig 8.2: Second Level RTL Schematic of ANC

8.3 Simulation Waveform of ANC(Start)

TGPCET/ECE/2015-16

Fig 8.3: Simulation Waveform of ANC(Start)

8.4 Simulation Waveform of ANC (End)

Fig 8.4: Simulation Waveform of ANC (End)

37

TGPCET/ECE/2015-16

8.5 Top Level RTL Schematic of ANC With Pipeline

Fig 6.5: Top Level RTL Schematic of ANC With Pipeline

8.6 Second Level RTL Schematic of ANC with Pipeline

Fig 8.6: Second Level RTL Schematic of ANC with Pipeline

38

TGPCET/ECE/2015-16

8.7 Simulation Waveform of ANC(Start) With Pipeline

Fig 8.7: Simulation Waveform of ANC(Start) With Pipeline

8.8 Simulation Waveform of ANC (End) With Pipeline

Fig 8.8: Simulation Waveform of ANC (End) With Pipeline

39

TGPCET/ECE/2015-16

8.9 Design Summary of ANC

Fig 8.9: Design Summary of ANC

8.10 Design Summary of ANC With Pipeline

Fig 8.10: Design Summary of ANC With Pipeline

40

TGPCET/ECE/2015-16

8.11 Timing Reports of ANC

Fig 8.11: Timing Reports of ANC

8.12 Timing Reports of ANC With Pipeline

Fig 8.12: Timing Reports of ANC With Pipeline

41

TGPCET/ECE/2015-16

8.13 Power Reports of ANC

Fig 8.13: Power Reports of ANC

8.14 Power Report of ANC With Pipeline

Fig 8.14: Power Report of ANC With Pipeline

42

TGPCET/ECE/2015-16

43

8.15 Comparison of parameters with and without Pipelining

Table 8.1: Comparison of parameters with and without Pipelining

PARAMETERS

WITHOUT
PIPELINE

WITH PIPELINE

Area

46%

61%

Dynamic Power

42.36mW

11.18 mW

TIME

217.639 ns

239.373 ns

CHAPTER IX
Conclusion & Suggested Further Work
In this report an efficient design for the implementation of an LMS
adaptive filter for adaptive echo cancellation is presented.

Typical digital

signal processing elements may be optimized along an area, power, timing

triangle.

Our design implements a pipelined architecture to achieve

efficiency. Synthesis and implementation results of the original 40 tap LMS

TGPCET/ECE/2015-16

44

filter and the pipelined 40 tap LMS filter show that the improved design
shows a reduction in power consumption of 20%.

However, due to the

pipelined design, area is increased by approximately 23%.

Future work on this design may be carried out to further fold the
pipeline design to achieve higher power efficiency at the cost of greater
latency. Another possible approach to improving power efficiency is to use
newer generation FPGAs based on smaller feature sizes as these provide
much lower static power consumption compared to previous generation
FPGAs such as the Spartan 3A DSP FPGA that this design was implemented
on.

REFERENCES
[1]

Pramod Kumar Meher, Sang Yoon Park -Adaptive-Delay-power Efficient

Fixed- point LMS Adaptive Filter With Low Adaptation Delay,

IEEE

transaction on very large scale integration system vol.22 no 2. feb

[2]

2014.
Ms. S. Savitha, Ms. S. Lakshmi, - Implementation Of Efficient LMS
Adaptive Filter with Low-Adaptation Delay, IEEE sponsored 2nd

TGPCET/ECE/2015-16

45

International Conference On Electronics And Communication Systems

[3]

(ICECS 2015).
Bhumika Chandrakar, O. P. Yadav and V. K. Chandra,- A survey Of
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[4]

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Pranjali M. Awachat A Design Approach For Noise Cancellation In
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[5]

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International Journal of computer and Electrical Engineering, Vol. 2,

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[10] D. Nicolae, R. Romulus, Noise canceling in Audio signal with

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with Variable step Size, Conference on Information Sciencs and

Systems, The Johns Hokins University, 2005.
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