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Anexo 2 - Plantilla IEEE
Anexo 2 - Plantilla IEEE
A continuación encontrará un archivo word IEEE editable para que puedan realizar los documentos de las
actividades del curso. Es de recordar que solo el abstract se escribirá en inglés.
Abstract — In this work it is presented a DSP card PhD Student, Ted Hoff. In this article it is presented eighteen
implementation method of the FxLMS single channel algorithm (18) offline results for LMS system identification algorithm,
for active noise control (ANC). The ANC technique uses the also it is presented twenty one (21) simulation results of the
superposition principle to attenuate acoustical noise. It is
FxLMS ANC algorithm, and finally the single channel
presented some simulating and experimental results of the LMS
and FxLMS algorithms. The signals used in the ANC are several
FxLMS algorithm it is implemented in Simulink to download
low frequency tones and a broadband sound signal (air it then to the DSP card, in order to obtain experimental real
compressor), which is a machine widely used in industrial time results. The simulations are important, because it can be
applications. The experiments are carried out in an enclosure test if the algorithm is working well and also to compare the
(control room of approximately 36 cubic meters of volume). performance of the algorithms between simulations and real
time results. ANC has been widely investigated for
Keywords—component; system identification, LMS algorithm, applications like, noise control in air ducts [13], headphones
FxLMS algorithm, active noise control, Matlab, Simulink, adaptive applications [14], aircraft cabins [17], multi-channel ANC for
filtering, FIR filters, secondary path estimation, transfer function,
automobiles [18], 3d reverberant enclosures multi-channel
adaptive algorithm, impulse response, FIT, white noise.
ANC [15], but single channel ANC in 3d reverberant
I. INTRODUCTION (HEADING 1) enclosure for single channel, has been little documented,
Acoustic noise affects the life quality of the society, causing always is used multichannel setups (multiple control sources
stress and issues related to ear. In most of industries, it exists and error sensors) to achieved considerable attenuation of
machines, which generate high sound pressure levels and can approximately 20 dB [16-17]
be considerate like acoustic noise sources. In order to minimize
the negative effects on employees, generally it is used passive This article is divided as follows: Section II, System
methods, like acoustic isolation with acoustic barriers (walls), description, Section III, Simulation results. Section IV,
and hearing protectors. These passive techniques are widely Implementation of the FxLMS algorithm for active noise
used, but its performance in low frequencies is poor [1]. control in the DSP Texas TSM320C6713DSK, Section V,
Conclusions, and Section VI, Future work.
Active noise control refers to attenuate acoustic noise at
specific space points, without using passive methods (physical II. SYSTEM DESCRIPTION
barriers or hearing protection), between the noise source and For ANC is commonly used two different configurations of
the receptor. This technique is based on the wave the FxLMS algorithm, one is a feedback ANC approach
superposition principle, which tells how waves may interfere proposed by Olson and May in 1953 [10], in this scheme, a
on a constructive or destructive way, and just adding to the microphone is used as error sensor and also as reference
existing acoustic field, another wave with the same frequency sensor.
and amplitude, but opposite phase, then they interfere
destructively and cancel each other out. For ANC algorithm
design purposes, it is used the LMS algorithm invented by the
Standford University Professor, Bernard Widrow and his first
Figure 2. LMS algorithm block diagram, for system
This control setup is commonly used for control narrow band identification
or predictable noises, one of the applications of this approach Where x(n) is the input to the system identification
is controlling the sound field in headphones and hearing algorithm, which generally is a signal with equal energy in all
protectors [11]. The second is a feed-forward ANC approach, the frequencies, in this case it is used white noise as the input
which uses two sensors, an error sensor and a reference sensor. of the system. P(z) is the system to identify, which is the
This setup is used for narrow band noise control using a non- electro acoustic system (enclosure and transductors). W(z) is
acoustic reference sensor “accelerometer”, and for broad band the adaptive filter, whose coefficients are updated by the LMS
noise control using an acoustic reference sensor ”microphone” adaptive algorithm, until the error signal converges to zero.
[7]. In this case, it will be used a microphone as the reference
For the secondary path estimation, it is necessary an input
sensor. In the figure 1, we can observe the FxLMS block
and an output, which are the white noise and the signal
diagram algorithm for feed-forward active noise control.
recorded by the error microphone. Then, it is possible to put
these audio signals (input-output) into the algorithm to identify
the coefficients of the impulse response.
The adaptive algorithm depends of an optimization criterion
or cost function, which for LMS, is based in minimize the
mean square error, as it is shown in equation 1. Where d(n) is
the desired signal (observed output), and y(n) is the predicted
output.
J=E ¿
In order to minimize the MSE, it’s used the descendent
gradient method, shown in the equation 2. The equation 3 is the
Figure 1. FxLMS Algorithm for Feed forward Active Noise
solution to the descendent gradient method which minimizes
Control. [2]
the cost function, and it is obtained assuming that the input and
The Primary noise is the undesirable signal, which is tried output signals are jointly wide sense stationary processes [12].
to be attenuate, and this is measured by the input microphone
(reference signal). The secondary source (cancelling speaker) h ( n+1 ) =h ( n )−μ ∇ J ( n ) (2)
generates the control signal and finally the primary noise is
attenuated at the physical error microphone position. h ( n+1 ) =h ( n ) +2 μe ( n ) x ( n ) (3)
It is important to know the software elements that are part Where h(n+1) is the coefficient at instant (n+1), μ is the
of the ANC controller, these are: the LMS adaptive algorithm step size which controls the convergence and the stability of the
which update the coefficients of the W(z) adaptive filter, which algorithm, e(n) is the error signal and x(n) is the input signal.
is this case is represented as a FIR filter [8]. The C(z) filter B. Analitical solution (Least Mean Square) algorithm for
represented the secondary path estimation or the transfer system identification
function between the secondary source (control source) and the
error microphone [2].
Let us consider de output filter as a FIR filter
A. LMS (Least Mean Square) algorithm for system
N−1
identification
y ( n )= ∑ h (k )x (n−k )= X (n)T h
In this part it is described the implementation procedure of the k=0
LMS algorithm for system identification, which will be useful
in active noise control to estimate the secondary path transfer
function. The figure 2, describes the LMS block diagram Where:
algorithm, for system identification. T
x ( n )=[ x ( n ) , x ( n−1 ) , x ( n−2 ) , … .. , x ( n−N +1 ) ]
.
h ( n )=[ h ( 0 ) , h ( 1 ) , h ( 2 ) , … .., h ( N−1 ) ]