+ (2)
Where
2
) , ( l k S the power spectrum of the clean
speech signal and
2
) , ( l k S is the power spectrum of
the noise signal. The proposed algorithm is summarized
[Dhanasekaran, 3(1): January, 2014] ISSN: 22779655
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[1822]
in the block diagram shown in Fig. 1. It is consists of
main components: input signal ,adaptive noise,
VSSNLMS, and sigmoid function with a priori SNR.
III. Proposed Noise Estimation And
Reduction Algorithm
The experimental work has been done on
Matlab. The noise reduction algorithm is based on the
STD of the noisy power spectrum in a time and
frequency dependent manner as follows:
= =
= =
L
l
K
k
f l k X
L
k x l k X
K
l x
1
2
1
2
1 , ) , (
1
) ( , ) , (
1
) (
(3)
, ) ( ) , (
1
) (
2
1
2
(
(
(



\

=
=
K
k
t
t
l x l k X
K
l v (4)
, ) ( ) , (
1
) (
2
1
2
(
(
(



\

=
=
L
l
f
f
k x l k X
L
k v (5)
, ) (
1
, ) (
1
1
2
1
2
=
= =
K
K
f f
L
l
t t
k v
K
l v
L
(6)
,
) (
) ( ,
) (
) (
2 2
= =
f
t
f
t
t
l
k v
k
l v
l
(7)
where ) ( 1 l x is the average noisy power spectrum in the
frequency bin, ) (k x f is the average noisy power
spectrum for the frame index, and
2
t
and
2
f
are the
assumed estimate of noise power. (7) Gives the ratio of
the (STD) for the noisy power spectrum in the time
frequency bin to its average. In the case of a region in
which a speech signal is strong, the STD ratio by (7) will
be high. The ratio is generally not high for a region
without a speech signal. Therefore, we can use the ratio
in (7) to determine speech presence or speechabsence in
the timefrequency bins [7]. Classification of speech
presence and speech absence in frames using an adaptive
sigmoid function based on a posteriori SNR Our method
uses an adaptive algorithm with a sigmoid function to
track the threshold and control the tradeoff between
speech distortion and residual noise:
(
+
=
)) ) ( .( 10 exp( 1
1
) (
t t
t
l
l
, (8)
Where ) (l
t
is the adaptive threshold using the sigmoid
function. We defined a control parameter
t..
This
threshold ) (l
t
is adaptive in the sense that it changes
depending on the control parameter
t
. The control
parameter
t
is derived from the linear function using the
a posteriori signal to noise ratio (SNR) in frame index
[9].
, ) ( .
off s t
l SNR + =
(9)
( )
,
2 , ) (
2 , ) , (
log . 10 ) (
2
2
(
(
(
(
(


\

=
k D norm
l k X norm
l SNR
(10)
Where
=
5
1
2 2
) , ( 5 1 ) (
l
l k X k D is the average of
the
2
) , ( l k X initial 5 frames during the period of the
first silence and norm is the Euclidean length of a vector.
, .
min max
SNR
s off
= (11)
,
min max
max min
SNR SNR
s
=
(12)
Where
s
is the slope of the
t
,
off
is the offset of the
t
. the more the a posteriori SNR increases, the more the
t
decreases. Simulation results show that an increase in the
t
parameter is good for noisy signals with a low SNR of
less than 5 dB, and that a decrease in
t
is good for noisy
signals with a relatively high SNR of greater than 15 dB.
We can thus control the tradeoff between speech
distortion and residual noise in the frame index using
t.
the adaptive threshold using the sigmoid function allows
for a tradeoff between speech distortion and residual
noise by controlling
t
[5],[6],[8] . If a speech signal is
present, the ) (l
t
calculated by (8) will be extremely
small (i.e., very close to 0). Otherwise, the value of
) (l
t
calculated by (8) will be approximately 1.
IV. Experimental Results And Discussion
This approach is based on decision tree
technique of ANN. This technique are using to
prune less effective neurons from net work and
[Dhanasekaran, 3(1): January, 2014] ISSN: 22779655
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[1822]
finally running systems with successful onl y
neuron. In different conditions different neuron
may give better results. The survival
chances of neurons depend upon its algorithm
efficiency in given circumstances. The step size
factor was varied depending upon the optimization
required between complexity and amount of noise
cancellation.
Figure 2 VSSNDLMS algorithm samples and output
In Noisy conditions of case t wo
VSSNDLMS algorithm was most efficient.
Remaining all algorithm were decision tree, as
they were far away from standard threshold
value.
V. Conclusion
In this approach we found out those in particular
noisy conditions most suitable algorithms can be sort out
using our design Tree Method. This method can also be
applied to none cancellation, signal estimation and
vibration cancellation problems means of MSE Obtained
0.2469 deviation from them when calculated. It is
observed that VSSNDLMS is better than the other
unwanted application for EANC of SONG input or
speech for no. of iteration 2000. The Threshold criteria
result in 0.001 is the least values amongst the all the
applied algorithm.
Hence Pruning of all other algorithm include the
selection of that algorithm an Sum + Val neuron. Next
good performing Lms itself for 2000 Iteration the Excuse
performance of the Lms is due to Linear nature of the
song or speech input noise.
Here this destination have tried to reduce such
week signals by using basic adaptive approach But in
real time applications which involves thousand of such
week signals and knowing the fact but noise , at any cost,
cannot be cancelled at hundred percent, this approach
cannot do us a good favor and keep this research topic
active in future work.
The future ambition or task to apply this novel
concept for kalman and winer filters family algorithm
simultaneously , where an it is been tried to implement it
for only outer noise removal automobiles and noise
cancellation is head phones, mobile, noise performance
can be upgraded by finding proper step factor and
threshold Value . the algorithm level up gradation of
threshold and decision Tree Criteria may improve
performance.
Adaptive designed signal processing is rapidly
growing Branch of Digital signal Processing (DSP) used
has a great significance in the design of adaptive systems
. The various s signal processing applications demands
for reduction in fraud off between mid adjustment and
convergence rate, taking realization of algorithm into
account. These in slope of improvement in existing
algorithm which reduces complexity and fulfill and our
stringent conditions for stability. This thesis dealt with
trans versal FIR adaptive filter, this is only one method
of digital filtering and other techniques such inflict
impulse response (IIR) of lattice filtering may move to
be effective in an noise cancellation application. The real
time EANC system can be implemented successfully
using the TITmsc 6713 DSK.
Hence this system can only cancel those echos
that fair within the length of Adaptive filter (FIR). There
is lot, which can be done in future for improvement on
the methods for acoustic noise cancellation the fixed
digital signal processing and particularly adaptive
filtering is vast and further research and development in
this area can be result in some improvement on the
methods studying in this dissertation.
VI. References
[1] Dayang Zhou and Victor E. DeBrunner Efficient
Adaptive VolterraFilters For Active Nonlinear Noise
Control With A Linear Secondary Path in IEEE
2004.
[2] S. F. Boll, Suppression of acoustic noise in speech
using spectral subtraction, IEEE Trans. on
Acoustics, Speech, and Signal Processing, vol. 27,
no. 2, pp. 113120, 1979.
[3] Lu.Yingwei and Narashiman Sundarjan
Performance Evaluation of a Sequential Minimal
Radial Baisi Function(RBF) Neural
Network Learning Algorithm in IEEE trans. On
Neural Networks, vol.9, No.2,March 21998.
[4] Lu Yingwei , N.Sundararajan, P.Saratchandran
Adaptive Nonliner Systems Identification Using
Minimal Radial Bsis Function Neural Networks in
IEEE 1996
[Dhanasekaran, 3(1): January, 2014] ISSN: 22779655
Impact Factor: 1.852
http: // www.ijesrt.com(C)International Journal of Engineering Sciences & Research Technology
[1822]
[5] Y. Ephraim and D. Malah, Speech enhancement
using a minimum meansquare error shorttime
spectral amplitude estimator, IEEE Trans. On
Acoustics, Speech, and Signal Processing, vol. 32,
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[6] R. Sundarrajan and C. L. Philipos, A noise
estimation algorithm for highly nonstationary
environments, Speech Communication, vol. 48, pp.
220231, 2006.
[7] S. J. Lee and S. H. Kim, Speech enhancement using
gain function of noisy power estimates and linear
regression, Proc. of IEEE/FBIT Int. Conf.
Frontiers in the Convergence of Bioscience and
Information Technologies, pp. 613616, October
2007.
[8] Cohen, Speech enhancement using a noncausal a
priori SNR estimator, IEEE Signal Processing
Letters, vol. 11, no. 9, pp. 725728, 2004.
[9] ITUT, Subjective test methodology for evaluating
speech communication systems that include noise
suppression algorithm, ITUT Recommendation, p.
835, 2003.
[10] S.M. Kuo and D.R.Morgan ,Active Noise Control
Algorithms and DSP Implementations. New York
:Wiley ,1996
[11] Debi Prasad Das ,Filtered s LMS Algorithm for
Multichannel Activeontrol Of Nonolinear Noise
Processes in IEEE Transcactions on SpeechAnd
Audio Processing , vol 14 ,no.5 Spetember 2006
[12] Debi Das and Ganpati Panda , Active Mitigation of
Nonlinear NoiseProcesses Using A Noval Filtereds
LMS Algorithm in IEEETransactions of Speech
and Audio Processing, vol. 12,No..3,May 2004.
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