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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
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the
advent
of
telephony
echoes
have
been
problem
in
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increased
during
recording
(encoding),
and
then
decreased
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].
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Characteristics
Accuracy
Dolby and
DYNAMIC
dbx noise
noise
MATLAB
VHDL
reduction
reduction
technique
technique
technique
technique
Average
Average
Good
Better
squared (RMS)
of noise
encode/decode
algorithm.
II noise
reduction
systems, DNR
is a playbackonly signal
processing
Timefrequency
domain using
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
quality.
Application
It has
It is use in
It is used
revolutionized
ECG system
It is used in mobile
MP3,discs,
to read the
phones, digital
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
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advantages
of
easy
implementation
and
low
computational
in
Noise
Cancellation
for
Speech
Enhancement,
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11
No.6, 2004.
In
this
paper
hard ware
the
Donor
ECG
in
Heart-Transplant
Electrocardiography,
CHAPTER III
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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
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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
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distort
the
fidelity
of
the
transmitted
signals,
making
the
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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
is
that
of
acoustic
echo
cancellation
for
conference-style
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CHAPTER IV
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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.
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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.
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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.
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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.
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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
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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
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only VHDL
methodology
band
is
a description language,
but
also
design
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
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the
application
30
specified
integrated
circuit
(ASIC)
representation
or
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
Logic
Devices.
It
is
known
for
inventing
the
Field
in
end
markets
such
as
communications,
industrial,
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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.
6.2 Challenges
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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.
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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.
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.
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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.
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CHAPTER VIII
Analysis of Result
8.1 Top Level RTL Schematic of ANC
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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
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filter and the pipelined 40 tap LMS filter show that the improved design
shows a reduction in power consumption of 20%.
REFERENCES
[1]
IEEE
2014.
Ms. S. Savitha, Ms. S. Lakshmi, - Implementation Of Efficient LMS
Adaptive Filter with Low-Adaptation Delay, IEEE sponsored 2nd
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45
(ICECS 2015).
Bhumika Chandrakar, O. P. Yadav and V. K. Chandra,- A survey Of
Noise Removing Technique for ECG Signals, International Journal of
Advanced Research in computer and communication Engineering vol.
[4]
[5]
[6]
[7]
[8]
In
Noise
Cancellation
for
Speech
Enhancement,
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