INTERNATIONAL JOURNAL OF INNOVATIVE RESEARCH EXPLORER ISSN NO: 2347-6060

A LOW FREQUENCY NOISE CANCELLATION VLSI

CIRCUIT DESIGN USING LMS ADAPTIVE FILTER FOR IN-
EAR HEADPHONES.
Manu M Rajan, Student, Second Year M.E (VLSI DESIGN) Marthandam College of Engineering and
Technology, Kuttakuzhi, Tamilnadu

R.Dinesh, Associate Professor, ECE, Marthandam College of Engineering and Technology, Kuttakuzhi Tamilnadu

Abstract – The noising of audio signal is key challenge problem in Audio Signal Processing. Since noise is
random process and varying every instant of time, noise is estimated at every instant to cancel from the original
signal. There are many schemes for noise cancellation but most effective scheme to accomplish noise cancellation is
to use feed forward filtered-x least mean square. Active Noise Cancellation (ANC) is achieved by introducing “anti
noise” wave through an appropriate array of secondary sources. The FxLMS algorithm implemented in Mat lab using
Xilinx system generator to achieve noise control.. These factors are used to demonstrate the efficiency of the
algorithm used. Then it is implemented in VLSI platform through Verilog coding using Xilinx

I. INTRODUCTION

Conventional active noise cancelling (ANC) head phones often perform well in reducing the low-frequency
noise and isolating the high-frequency noise by earmuffs passively. The existing ANC systems often use high-speed
digital signal processors to cancel out disturbing noise, which results in high power consumption for a commercial
ANC headphone. The contribution of this paper can be classified into: 1) proper filter length selection; 2) low-power
storage mechanism for convolution operation; and 3) high-through put pipelining architecture. The existing active
noise cancelling (ANC) systems use delayed LMS adaptive filter. These ANC systems require high computational
complexity, power intensive hardware, and significant processing time for measuring noise signal, and then
calculating and synthesizing proper anti noise signals to cancel out the noise signals in real time

In this paper, a dedicated feed forward ANC circuit implementation based on the well-known FxLMS adaptive
algorithm for high fidelity in-ear headphones is developed. . By selecting a proper filter length, 62.7% of the
computation complexity of convolution filter can be saved without sacrificing any ANC performance. In addition, a
dedicated storage mechanism called one-update circular buffer is proposed to keep the switching activity low to save
power. Finally, a three-stage pipelining multiply accumulator (MAC) architecture is used to increase the data
throughput.

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INTERNATIONAL JOURNAL OF INNOVATIVE RESEARCH EXPLORER ISSN NO: 2347-6060

II SURVEY OF EXISTING METHODS

S. M. Kuo etal explained that Active noise control (ANC) is achieved by introducing a cancelling “anti noise”
wave through an appropriate array of secondary sources. These secondary sources are interconnected through an
electronic system using a specific signal processing algorithm for the particular cancellation scheme

I. Panahi, etal, explained the Active noise cancellation (ANC) applications for headphones are strongly influenced
by practical constraints. Most previous works developing algorithms for ANC headphones are based on simplified
simulations only and neglect practical limitations.

L. Wu, X. Qiu,etal mentioned in the adaptive feedback active noise control system based on the internal model
control (IMC) structure, the reference signal is regenerated by synthesizing the error signal and the secondary signal
filtered with the estimation of the secondary path, hence more computation load and extra programming are required

III EXISTING MODEL: DELAYED LMS ADAPTIVE FILTER

Fig 1: Modified delayed LMS adaptive filter

In the design based on pipelined with m pipeline stages, the error en will produced after m adaptation delay
cycles.So the algorithm therefore uses the en−m

. The DLMS adaptive filter weight-update equation is represented by

The DLMS adaptive filter block diagram is shown in Fig. 1, where the adaptation delay of m cycles amounts to the
delay introduced by the full adaptive filter structure and the structure consisting of finite impulse response (FIR)
filtering and the weight-update process.

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INTERNATIONAL JOURNAL OF INNOVATIVE RESEARCH EXPLORER ISSN NO: 2347-6060

The weight-update equation of the modified DLMS algorithm is given

by
The modified DLMS algorithm decouples computations of the error-computation block and the weight-update block
. It permitted us to perform optimal pipelining by feed forward cut-set retiming to minimize the number of pipeline
stages and adaptation delay.

IV PROPOSED MODEL: ADAPTIVE FEED FORWARD FILTERED-X LEAST MEAN

SQUARE FX LMS FILTER
A feed forward ANC control system uses an input microphone close to the noise source to pick up the noise
signal x(n) before it is sensed by the listener. Accordingly, the ANC controller can produce an anti noise signal
y(n)processing equal amplitude but opposite phase of x(n). Such anti noise signal is used to drive the cancelling-
loud speaker to generate a cancelling sound that attenuates the primary acoustic noise in the ANC system. Fig. 1
shows the application of the feed forward Fx LMS adaptive algorithm in ANC system

Fig2: Feed forward Fx LMS filter

P(z) and S(z), respectively, denote the primary-path and the secondary-path models. W(z) indicates the filter

weights of the ANC controller to adaptively generate the required anti noise signal according to the time-variant
noise source. and e(n) is the error signal generated by acoustically combining the primary noise d(n)and the adaptive
filter output y(n).

The output y(n) of an adaptive filter at time n is given by y(n) = w(n)x(n) Where w(n) = [w0(n)w1(n) . . .wL−1(n)]T
is the L × 1 filter coefficient vector and
x(n) =[x(n)x(n − 1) . . . x(n − L + 1)]T is the L × 1 reference signal vector. The FxLMS algorithm updates the
weighting coefficients of the adaptive filter in a way as w(n + 1) = w(n) + μ1x(n) ∗ h(n)e(n)

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where μ1 is the step size of the algorithm that determines the stability and convergence speed of the algorithm h(n)
is the impulse response of S(z) and
e(n) = d(n) − y(n).

The input vector x(n) is filtered by S(z) before updating the weighting vector. However, in practical
applications, S(z) is unknown and must be estimated by S(z). Therefore, the resulting FxLMS algorithm can be
represented by wl (n+1) = wl (n) + μ1x(n − l)e(n), l = 0, 1, 2, . . . , L − 1where x(n) =m(n)x(n − m), m = 0, 1, 2, . . . ,
M −1 , estimated version of the reference input x(n) after passing through the secondary path

Usually, the standard transversal filter used in the ANC controller .A standard adaptation algorithms proposed for
the transversal filters e.g. Least Mean Square (LMS) or Recursive Least Square (RLS) cannot be used for the
automatic adjustment of ANC controllers

FxLMS is a gradient based algorithm used for the identification of an unknown system (e.g. a desired ANC
controller) at the presence of a secondary path. The performance of an FxLMS-based ANC system is limited by a
number of related factors that must be addressed in the appropriate order. The absolute maximum level of
performance is limited first by the characteristics of the physical plant to be controlled, including the secondary path
impulse response and the acoustic noise bandwidth.

This means that no matter how good is the electronic control system, the FxLMS will not function properly if the
secondary path has a long impulse response and/or the acoustic noise has a wide bandwidth

V. RESULTS

Fig.3 output for first 2 clocks

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INTERNATIONAL JOURNAL OF INNOVATIVE RESEARCH EXPLORER ISSN NO: 2347-6060

Fig 4 : Final Output and input with noise

Table I Comparison with Existing works
SL. Parameters Existing Proposed
No methods method
1 Power high low
consumptio
n
2 Power high low
storage
3 Algorithm Delayed LMS Fast LMS
Algorithm Algorithm
4 cost for high low
Headphone
s
5 Size and large : Small
weight of Because of
Headphone additional use
of Battery
6 Mathemati less in less in
cal Secondary Secondary
complexity path path

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From the table, it is clear that power consumption is low compare to the previous works and the headphones
cost is also low
VI. CONCLUSIONS

In this paper, an area-/power-efficient feed forward Fx LMS ANC circuit has been developed for
in-ear head phones. The proposed design has been successfully implemented by using TSMC 90-nm CMOS
technology. To verify the validity of the proposed design, a series of physical measurements has been executed in an
anechoic chamber. Furthermore, the measurement results were compared with that of other state-of-the-art works.
The experimental results show that the proposed high-performance/low-power ANC circuit design can reduce
disturbing noise of various frequency bands very well, and outperform the existing works. The proposed design can
attenuate 15 dB in physical measurement for the broad band pink noise between 50 and 1500 Hz when operated at
20-MHz clock frequency at the cost of 84.2 k gates and a power consumption of 6.59 mW only.
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