International Journal of Scientific Research Engineering & Technology (IJSRET), ISSN 2278 0882

Volume 3, Issue 8, November 2014

1258

Hardware Implementation of Speech Enhancement Algorithms

Kuldeep Sharma1, Bhumika Bisht2, Pankaj Goel3
1, 2, 3
IMS Engineering College Ghaziabad UP

Abstract
This paper implements a VLSI based Adaptive noise
canceller (ANC) using Least Mean-Square (LMS) and
Normalized Least Mean-Square (NLMS). Adaptive filters
are capable of adapting their filter coefficients as per the
variations in characteristics of input signal and noise to
achieve a noise free signal. These algorithms have been
implemented on Digital Signal Processor. Platform used for
implementation of algorithms is DSP processor
TMS320C6713.

Primary
Signal

d
Reference Signal

Adaptive Filter
-

Figure1. Adaptive Noise Canceller

Keywords- ANC; LMS; NLMS; DSP Processor

III.
I.

In many applications of digital signal processing,

Adaptive digital filters have advantages over the traditional
filters. Adaptive filter have wide range of applications
including noise cancellation, channel equalization in
telecommunication system & system performance etc.
Adaptive filters are self modifying digital filter that adjust its
coefficient in order to minimize error function.
FIR filters are employed in filtering problem where there
is requirement for a linear phase characteristic within the
passband of filter. The adaptive FIR filtering adopts
weighted summation process of input sequence. LMS
algorithm is most popular stochastic gradient algorithm [1],
[2].The convergence rate & stability of the LMS filters can
be recursively governed by proper choice of the step size as
a closed loop control system. The step size must be reduced
in direct proportion to the length of adaptive filters [3]. The
LMS filter architecture is examined in terms of the speed,
power consumption & resource used.

II.

LMS ALGORITHM

INTRODUCTION
LMS algorithm is most general & simple form of
filtering. The LMS shows advantage of less computational
complexity & simple implementation [4]. LMS involves the
initial estimation of coefficient vector and updating in a
particular direction to achieve the desired response. At each
point the filter changes its coefficient from present state to
next state of coefficient vector. The filter coefficients are
adjusted such that the mean square error (MSE) should be
minimized with the objective of minimum of function
[5].The coefficient of the adaptive filter at minimum error
occurs at W = W0, represents that the current filter
coefficients equal the optimal coefficient values. Optimum
weight of filter can be given as
0 =

where auto correlation of the reference signal & is

the cross correlation of the reference and the primary signal.
The mean square error is given by
1
= 2

ADAPTIVE NOISE CANCELLER (ANC)

The adaptive filer is used to cancel unknown interference

contained in primary signal, with the cancelation being
optimized in some sense. The primary signal served as the
desired response for the adaptive filter. A reference signal
derived from sensors or set of sensors, is employed at input
to adaptive filter.

(1)

(2)

where 2 is the variance of the desired signal.

But in the LMS algorithm filter coefficient moves in
particular direction with each iteration. The direction of the
filter is such that the error can be reduced. The new value of
can be determined by iteration is given by-

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International Journal of Scientific Research Engineering & Technology (IJSRET), ISSN 2278 0882
Volume 3, Issue 8, November 2014

+ 1 = J

(3)

where J is the gradient of the mean square error, n is time

index & is the step size. The step size is crucial factor in
the algorithm performance & implementation. It determines
the convergence rate, asymptotic error for algorithm [6]. The
coefficient update for the LMS algorithm, which can be
expressed as
+ 1 = + 2

(4)

From above equation, the filter coefficient is updated

iteratively in a straightforward method. The second term in
equation (4) is the correction factor. The step size, which
determine the amount by which the coefficients are allowed
to change, e(n) current error. In adaptive filter the N sample
of the reference input is given to the FIR filter with filter
coefficient W. The output of filter y(n) is subtracted from
the primary signal d(n).The result shows error signal, which
is output of algorithm. This output is given to input of LMS
to update the weight for the next iteration. Output of filter is
given by
=

(5)

(6)

IV.

1259

EXPERIMENTAL SETUP & RESULT

Digital Signal Processor (DSP) TMS320C6713 has been

used for the Real Time processing. The TMS320C67XX
DSPs compose the floating-point DSP generation in the
TMS320C6000 DSP platform. The C6713 device is based
on the high-performance, advanced very-long-instructionword (VLIW) architecture developed by Texas Instruments
(TI), making this DSP an excellent choice for multichannel
and multifunction applications. The C6713 has a rich
peripheral set that includes two Multichannel Audio Serial
Ports. Code Composer Studio is an integrated development
environment (IDE) that supports TI's Microcontroller and
Embedded Processors portfolio.
The algorithm used for the processing is LMS & NLMS.
The code for the algorithm has been implemented in C. The
experimental setup for the implementation is shown in fig 2.

& error

With the terms of equation (5) & (6) equation (4) is used to
represent the equation of LMS algorithm [7]. The boundary
of operation for the adaptive filters depends on the step size
for the algorithm to be stable the operation boundary is
given by
2
0<<
,
(7)

where is the largest eigenvalue of autocorrelation of

the input signal [8].
NLMS ALGORITHM
The NLMS algorithm is an improvement of LMS
algorithm. The only difference is in the weight updation
portion. Here step size is normalized by autocorrelation
function R.
() = (). ()
() = () ()
( + 1) = () +

.().()
()()

(8)
(9)
(10)

As the NLMS algorithm uses a variable step size in each

iteration NLMS is known to give better convergence
characteristics than the LMS. [11]

Figure 2: Experimental Setup

Two tests have been done. The LMS & NLMS has been
implemented. The white noise is used for the mixing with
the noisy signal. The noisy signal is mixed with white
Gaussian noise. The filtered output from DSK is recorded
with the Goldwave software. The recorded signal is further
processed in MATLAB to calculate the SNR.
The SNR of noisy signal before the filtering is 11.6323
dB. Fig 3 shows the filtered output from DSK in case of
LMS. After the filtering SNR is 15.9174 dB. The output
shows that SNR improvement is found.

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International Journal of Scientific Research Engineering & Technology (IJSRET), ISSN 2278 0882
Volume 3, Issue 8, November 2014

1260

REFERENCES

Figure 3: Filtered output from DSK

Table I: Analysis of SNR improvement
Sr.
No.
1
2
3

Name

SNR in dB

Noisy Signal
Filtered Signal for LMS
Filtered Signal for NLMS

11.6323 dB
15.9174 dB
15.9924 dB

As shown in table the SNR improvement for the NLMS is

also obtained in similar fashion. The proposed system is
capable of removing the noise from the noisy signal. For the
improvement the ANC for the LMS & NLMS is designed &
analyzed. The result is shown in table I. From the result in
table it can be analyzed NLMS ANC removes more noise
compared to the LMS.
The results that obtained for the designed system shows a
good performance for the system and considerable
improvement of signal to noise ratio was achieved.

V.

CONCLUSION

The implementation of Adaptive Noise Cancellation

(ANC) system on DSK is done successfully. SNR
improvement is achieved in both cases. When the results of
two algorithms were compared it is proved that the NLMS
algorithm has the best performance. The preliminary results
show that the system is working successfully though there is
definitely a scope for further improvement. Algorithms like
Block LMS and RLS can be implemented on this platform
which may perform better than LMS and NLMS.

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