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C

OLOURED
N
OISE
R
EMOVAL

AND

E
QUALISING

THE
C
HANNEL

EFFECT

FROM

A

NOISY

AUDIO

SIGNAL

G. Anudeep Reddy (EC08484)
G. Madhuri (EC08485)
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INTRODUCTION
We want to transmit a song signal.
Song Signal Transmitter
Receiver Loud Speaker
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REMOVAL OF NOISE
The signal may be corrupted by Noise.
Song
Signal
Transmitter
Receiver
Noise
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REMOVAL OF NOISE
The signal may be corrupted by Noise.
Song
Signal
Transmitter
Receiver
Loud
Speaker
Noise
Filter
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WHITE NOISE
White noise is a signal (or process), having
equal power in any band of a given bandwidth
(power spectral density).
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COLORED NOISE
Based on Spectral density (power distribution
in the frequency spectrum) we can distinguish
different types of noise.

This classification by spectral density is given
"color" terminology, with different types
named after different colors.
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COLORED NOISE
The color names for these different types of
sounds are derived from a loose analogy
between the spectrum of frequencies of sound
wave present in the sound and the equivalent
spectrum of light wave frequencies.

That is, if the sound wave pattern of "blue
noise" were translated into light waves, the
resulting light would be blue, and so on.
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PINK NOISE
Similar to White Noise except the power
density decreases 3 dB per octave as the frequency
increases. In technical terms the
density is inversely proportional to the frequency.
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BLUE NOISE
Similar to White Noise except the power
density increases 3 dB per octave as the frequency
increases. In technical terms the
density is proportional to the frequency.
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GENERATION OF COLORED NOISE
Colored noise can be generated by passing the
white noise through a shaping filter.
The response of the colored noise can be
varied by adjusting the parameters of the
shaping filter.


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White
Noise
Shaping
Filter
Colored
Noise
REMOVAL OF CHANNEL EFFECT
The signal may be corrupted by channel
transfer function also.
Song
Signal
Transmitter
Receiver
Noise Channel
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REMOVAL OF CHANNEL EFFECT
The signal may be corrupted by channel
transfer function also.
Song
Signal
Transmitter
Receiver
Loud
Speaker
Noise
Filter
Channel
Equalizer
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EQUALIZER
The equalizer should have transfer function
which is inverse of channel.


Where H(z) is the transfer function of equalizer
and C(z) is the transfer function of channel.
But in most of the cases we do not know the
transfer function of the channel, so we will
adapt the equalizer transfer function using
Learning algorithm.
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) (
1
) (
z C
z H =
COMPLETE BLOCK DIAGRAM
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CHANNEL AND CHANNEL EQUALIZER
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o A 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.

o This is in contrast to infinite impulse response (IIR)
filters, which have internal feedback and may continue
to respond indefinitely (usually decaying).

o The impulse response of an Nth-order discrete-time
FIR filter, lasts for N+1 samples, and then dies to zero.
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o FIR filters can be discrete
time or continuous-time, and digital or analog.
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For a discrete-time FIR filter, the output is a
weighted sum of the current and a finite number of
previous values of the input. The operation is
described by the following equation, which defines the
output sequence y[n] in terms of its input
sequence x[n]:



o The channel can be considered as a discrete
time digital FIR filter


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From the block diagram, it is evident that the optimal
equalizer should have transfer function which is inverse
of channel. Hence channel equalization is also known as
inverse filtering.

Transfer function of Channel , C(z)= b
0
+ b
1
z
-1
+ b
2
z
-2
+ b
n
z
-n


Transfer function of the Equalizer, H(z)= 1/C(z)
o Similarly the equalizer can be considered as an FIR
filter(discrete time, digital FIR filter)
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AN ADAPTIVE LINEAR EQUALIZER
Z
-1
Z
-1
Z
-1




x
k

x
k-1

x
k-2
x
k-L+1

w
0k
w
1k

w
(L-1)k

y
k

w
2k

There is an input signal vector,
a corresponding set of adjustable weights,
a summing unit, and a single out put signal.
1 1 , 0
...
L
x x x
1 1 0
... ,
L
w w w
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ADAPTIVE LINEAR EQUALIZER
o A procedure for adjusting or adopting the weights is
called weight adjustment or adaptation procedure.
o The combiner is called linear because for fix setting of
weights its output is a linear combination of the input
components.
o The output of the combiner can be represented as
l k
L
l
lk k
x w y

=

=
0
where denotes weight at instant.
lk
w
th
l
th
k
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If the weight and input vectors are expressed as

T
K L k k k
x x x X ] [
) 1 ( 1 0
=
T
K L k k k
w w w W ] [
) 1 ( 1 0
=
then the output is given by
k
T
k k
w x y =
The weights of the combiner are to be updated
using various learning algorithms
Ravi Kumar Jatoth Department
of ECE NITW
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LEARNING ALGORITHMS

LMS Algorithm
RLS Algorithm
Kalman Filter
Neural Algorithm
Fuzzy Logic System
Optimization Algorithms

All the algorithms update the weights of the
equalizer using different cost functions.
Ravi Kumar Jatoth Department
of ECE NITW
LEAST MEAN SQUARE ALGORITHM
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LMS: adaptive filtering algorithm having
two basic processes
Filtering process, producing
1) output signal
2) estimation error
Adaptive process, i.e., automatic
adjustment of filter tap weights

LEAST MEAN SQUARE ALGORITHM
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o LMS algorithm is one of the conventional
techniques applied to channel equalization. The cost
function is Mean Square Error (MSE). It updates the
weights of the adaptive FIR filter based on the error
obtained. The instantaneous error at any time-step 'k'
can be represented as

e(k) = d(k) y(k)

where d(k) delayed input reference is signal at time-
step k, and 'y(k) is estimated output from equalizer.
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o The equalizer filter's impulse response vector is
adapted using the following equation,

w(k+1) = w(k) + 2.e(k).x(k)
where is called Convergence factor or Learning
rate parameter, (0 1).
x(k) Is input from transmitter at time-step 'k'.

o This procedure is repeated till the Mean Square Error
(MSE) of the network approaches a minimum value.

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351M Digital Signal Processing
STABILITY OF LMS
More practical test for stability is




Larger values for step size
Increases adaptation rate (faster adaptation)
Increases residual mean-squared error

power signal input
2
0 < <
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Fig. 2 Adaptive filter using LMS algorithm
Z
-1
Z
-1
Z
-1



LMS
Algorithm

x
k

x
k-1

x
k-2
x
k-L+1

w
0k
w
1k

w
(L-1)k


e
k

y
k

d
k

-
+
w
2k

| |
T
1 1 k k k k L
x x x
+
= X
the L-by-1 tap input vector.
( )
T
0 1 1 k k k L k
w w w

(
=

W
the L-by-1 tap weight vector
LMS BLOCK DIAGRAM
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Z
-1

Z
-1


Z
-1

Z
-1


LMS
+
-
y (k)
e (k)
a
0

a
1

a
2

a
3

a
4

d(k)
x(k)
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NUMERICAL EXAMPLE- CHANNEL EQUALIZATION
Transmitted signal: random sequence of
1s.
The transmitted signal is corrupted by a
channel.
Channel impulse response:
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The amplitude distortion, and eigen value spread,
were controlled by W.

The received signal is processed by a linear, 11-tap
FIR equalizer adapted with the LMS algorithm

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REFERENCES
Digital Signal Processing using MATLAB demos
by Charulatha Devi.

Georgi Illiev and Nikola Kasabov, "Channel
Equalization using Adaptive Filtering with
Averaging", University of Otago, Newzeland.

M Reuter, J Zedlier, "Nonlinear effects in LMS
adaptive equalizers", IEEE Trans.Signal
Processing, June1999.
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