Baltimore, Maryland

NOISE-CON 2010
2010 April 19-21
Anovel delayless frequency domain filtered-x least mean square
algorithm for vehicle powertrain noise control
Jie Duan
a)
Teik C. Lim
b)
Vibro-Acoustics and Sound Quality Research Laboratory
Department of Mechanical Engineering, P.O. Box 210072
University of Cincinnati
Cincinnati, OH 45221-0072
A novel overlap-save, delayless frequency domain least mean square (LMS) algorithm is
formulated for treatment of vehicle powertrain noise. The proposed algorithm combines
the delayless implementation that eliminates the undesirable block delay in the signal path
while still retain all the advantages of the frequency domain approach, and the overlap-
save concept that overcomes the inherent slow convergence of conventional delayless
scheme when applied to control harmonic response. Numerical simulations applying the
proposed overlap-save delayless approach show substantial improvement in active noise
control performance compared to the direct frequency domain LMS algorithm.
a)
Email address: duanje@mail.uc.edu
b)
Email address: teik.lim@uc.edu
1 INTRODUCTION
Active noise control (ANC) is a technique based on introducing a controllable secondary
sound source that is out-of-phase from the original (primary) response, such that the resultant
sound is suppressed or significantly diminished
1,2
. Recently, a frequency domain implementation
of the least mean square (LMS) algorithm was introduced for ANC applications. This concept
possesses two major advantages: (i) the computational complexity is reduced by using the fast
Fourier transform (FFT); and (ii) faster convergence is possible because the step size can be
adjusted in each frequency bin
3-6
.
However, the traditional frequency domain control algorithm adds an additional block delay
in the signal path since response signal is processed block by block. This inherent delay critically
limits the performance of the control system especially for response with time-varying
frequencies, such as vehicle powertrain noise where the orders change with the engine crank
speed. To avoid this undesired delay, a delayless frequency domain LMS (DLMS) algorithm was
developed as discussed in Refs. 7 and 8.
In the DLMS algorithm, the adaptive weights are updated in the frequency domain, but
transformed into an equivalent set of time domain filter coefficients using the inverse FFT. Thus,
the control filter is implemented in the time domain. In this way, the inherent delay can be
avoided. However, the convergence speed in this basic delayless concept may not be sufficient
when applied to engine speed sweep condition. Therefore, the goal of the present work is to
develop further enhancements to the DLMS algorithm and examine the resulting effectiveness in
treating vehicle powertrain noise.
In this paper, to determine the limitations of the DLMS algorithm, its performance is first
studied using a set of purely tonal signals. The numerical simulations reveal that the convergence
of the DLMS algorithm highly depends on the frequency of the targeted tonal signal, even
though the secondary path is a simple pure delay one. This property may cause significant
degradation in the overall performance of the active control system for time-varying signal as
mentioned above, especially when convergence is unacceptably slow. This limitation cannot be
simply compensated by adjusting the step size for each frequency bin. To overcome the
drawback, we proposed an overlap-save concept that is further tested on actual measurements of
the powertrain noise in the passenger compartment.
2 CONVENTIONAL DLMS ALGORITHM
The delayless frequency domain LMS algorithm is attractive because it eliminates the
undesired block delay in the signal path
7,8
. The adaptive weights of the DLMS are computed in
the frequency domain like a traditional frequency domain algorithm. However, unlike the
traditional frequency domain algorithm, the adaptive weights are then transformed by inverse
FFT to update the filter coefficients in time domain. This allows the adaptive filter to process the
data sample by sample directly in the time domain.
Figure 1 shows the block diagram of the delayless frequency domain LMS algorithm for
ANC application. The system is designed to reshape the sound spectrum rather than simple
suppress the response amplitude. Therefore, the classical error signal is replaced by a pseudo-
error signal
'
( ) e n defined as
'( ) ( ) ( ) e n e n d n = (1)
where ( ) e n is the residual error signal sensed by the error microphone and is the difference
between the primary disturbance (uncontrolled response) ( ) p n and the secondary sound that is
the control signal y(n) filtered by the secondary path (transfer function between the control
source and response point of interest) S(z). Also, in the equation (1), ( ) d n is the desired sound
pressure, which is synthesized according to a certain pre-determined sound quality criteria.
In the proposed algorithm, one block of reference signal data and one block of pseudo-error
signal data are accumulated in the buffer separately to generate two vectors with N-point signal
data, namely ( ) x k and
'
( ) e k given by
( ) [ ( ) ( 1) ( 1)]
T
x k x kN N x kN N x kN = + (2)
' ' ' '
( ) [ ( ) ( 1) ( 1)]
T
e k e kN N e kN N e kN = + (3)
where k is the block index, N is the block length that is equal to the adaptive filter length, and
( )
T
denotes the transpose operation. The above two vectors ( ) x k and
'
( ) e k are transformed once
every N samples by an N-point FFT to produce a pair of frequency domain vectors expressed as
0 1 1
( ) { ( )} [ ( ) ( ) ( )]
T
N
X k FFT x k X k X k X k

= = (4)
' ' ' '
0 1 1
'( ) { ( )} [ ( ) ( ) ( )]
T
N
E k FFT e k E k E k E k

= = (5)
Similar to the filtered-x least mean square (FxLMS) algorithm, to avoid the distortion
caused by the secondary path, filtered reference signal is required and calculated in frequency
domain, which can be represented as
'
( ) ( ) , 0,1,..., 1 m
m m
X k X k S m N
.
= =
(6)
where m S
.
is the estimated impulse response of the secondary path in frequency domain and m is
the frequency bin index. It follows that the update strategy for the filter weights is the complex
least mean square algorithm (CLMS)
1
given by
( 1) ( ) '( ) '( ) W k W k X k E k + = + (7)
where ( ) denotes the conjugate operation and is the convergent factor. Again, can be
adjusted for each frequency bin. The adaptive filter weights in frequency domain are updated
every N samples. Thus, the weights vector of adaptive filter in time domain is also updated every
N samples using the inverse FFT operation:
{ }
0 1 1
( ) ( ) [ ( ) ( ) ( )]
T
N
w k IFFT W k w k w k w k

= = (8)
Hence, the output of the adaptive filter y(n), which is used to drive the secondary control
speaker, is given by
( ) ( ) ( )
T
y n w k n = x (9)
where ( ) n x is the input vector defined as ( ) [ ( ) ( 1) ( 1)]
T
n x n x n x n N = + x .
3 OVERLAP-SAVE IMPLEMENTATION
In this section, an overlap-save implementation of the delayless frequency domain LMS
algorithm for ANC applications is proposed as shown in Fig. 2. The differences of the proposed
DLMS from the conventional DLMS algorithm in the last section are discussed as follows. Here,
the reference signal comprises of the current block of data and another N samples from the
previous block, while the error signal is padded with N zero samples to yield 2N vector length
9-12
.
Therefore, equations (2) and (3) can be rewritten as
( ) [ ( ) ( 1) ( 1) ( ) ( 1)]
T
x k x kN N x kN N x kN x kN x kN N = + + (10)
'( ) [0 0 0 '( ) '( 1) '( 1)]
T
e k e kN e kN e kN N = + + (11)
These two signal vectors are then transformed into the frequency domain by applying the 2N-
point FFT routine. The resultant frequency spectra are given by
0 1 2 1
( ) { ( )} [ ( ) ( ) ( )]
T
N
X k FFT x k X k X k X k

= = (12)
' ' '
0 0 2 1
'( ) { '( )} [ ( ) ( ) ( )]
T
N
E k FFT e k E k E k E k

= = (13)
The filtered reference signal '( ) X k can then be computed by applying the same operation
demonstrated in equation (6) to ( ) X k given in the above equation. To ensure strict linear
correlation between the filtered reference and error signals, a process called gradient constraint is
applied. As shown in Fig. 2, this process is grouped by three functional blocks, namely IFFT,
zero last N point and FFT. The block gradient estimate is calculated as follows
11
:
{ }
( ) '( ) '( ) k first N terms of IFFT X k E k
.
V =
(14)
where '( ) X k is the conjugate of '( ) X k . The algorithm then transforms the time domain gradient
back into its frequency domain counterpart after forcing the last N value of the time domain
gradient to zero and updating the adaptive filter weights as shown in the following equation,
( 1) ( ) ( ), 0, 0, , 0
T
T
W k W k FFT k
.

+ = + V
`
)
(15)
It may be noted that the frequency domain adaptive weight vector W(k) in the modified
DLMS algorithm has 2N components as compared to the conventional one in equation (7) with
only N components. Thus, the weight vector w(k), previously computed from equation (8), of the
modified DLMS algorithm in time domain also possesses 2N components. The final N length of
the adaptive weights w(k), which is used to determine the input signal of the control speaker
according to equation (9), can be obtained by discarding the last N components,
{ } ( ) ( ) w k first N terms of IFFT W k = (16)
The resultant N length of the adaptive filter weights w(k) is used to produce the output signal y(n)
in time domain.
4 COMPUTER SIMULATION
Several numerical simulations are performed to examine the improvements and limitations
of the proposed modifications made to the basic DLMS algorithm. The first case is conducted by
using a pure tone signal to evaluate the performance of the conventional delayless frequency
domain LMS algorithm given in Fig. 1 and the proposed overlap-save implementation of
delayless frequency domain LMS algorithm as shown in Fig. 2.
The primary noise response is a pure tone at 100Hz or 110Hz. The reference signal is the
same sinusoid of unit amplitude with arbitrary initial phase. The sampling rate is 4096 and there
are 10,000 total samples. The block sizes for these two DLMS algorithms are the same and set to
128. The secondary path is a pure delay defined as
5
( ) S z z

= . Furthermore, the step size is

=0.0001 for all four simulations.
As shown in Fig. 3(a), the conventional delayless frequency domain LMS algorithm
converges very rapidly when the frequency of the primary signal is 100Hz. On the other hand, in
Fig. 3(b), the algorithm does not converge well for the frequency at 110Hz no matter what the
step size is selected. The reason is because the conventional delayless frequency domain
algorithm implements circular correlation as mentioned earlier. However, the modified delayless
frequency domain LMS algorithm given by Fig. 2 does not experience this problem. This is
clearly depicted in Fig. 4 (a,b) that demonstrates nearly the same convergence speeds in both the
100Hz and 110Hz tones. The same trends are seen in other frequencies even though the results
are not explicitly shown here.
The second case studied in this paper involves active control of powertrain noise when the
engine crankshaft speed is ramping up
13
. The baseline (primary) powertrain noise and the
corresponding tachometer signal (engine speed) were recorded in an actual experimental vehicle
with a 6-cylinder engine. The tachometer signal is used to generate the reference signal for the
control systems. A transfer function of the secondary path, which is from the control input
speaker to the microphone response location, was also obtained experimentally as well, as shown
in Fig. 5. For demonstration purpose, the objective of the following simulations is to reduce the
3
rd
order response, which is the main firing order, as much as possible.
Comparison ANC results for the different algorithms of interest are shown in Fig. 6. The
step size for each implementation is chosen to be the largest value while still assuring that the
ANC algorithms remain stable. In Fig. 6, the black line is the baseline response while the blue
line presents the control result by utilizing the fast least mean square (FLMS) algorithm in Ref. 6.
One can see very limited attenuation is attained. This is because the FLMS suffers inherent block
delay causing difficulty in tracking the harmonic frequency of powertrain noise precisely. In
addition, the green line shows the performance of the conventional delayless frequency domain
implementation of Fig. 1. It can be seen that the reduction varies for different engine speed. This
is due to slow convergent speed at some frequencies. In contrast, the enhanced delayless
frequency domain LMS algorithm shown in Fig. 2 yields much better performance, given by red
line in Fig. 6, as compared to the others.
5 CONCLUSIONS
An overlap-save implementation of the delayless frequency domain LMS algorithm for
active noise control of harmonic signals is successfully applied to time-varying powertrain
harmonics. Compared to the fast least mean square algorithm and conventional delayless
frequency domain least mean square algorithm, the modified delayless frequency domain
algorithm yields much better performance. Further study will be performed to extend the
proposed approach to multi-channel delayless frequency domain LMS algorithm.
6 REFERENCES
1. S. M. Kuo and D. R. Morgan, Active Noise Control Systems: Algorithm and DSP
Implementations, Wiley, New York, (1996).
2. S. J. Elliott, Signal Processing for Active Control, Academic Press, London, UK, (2001).
3. S. M. Kuo, M. Tahernezhadi and L. Ji, Frequency-domain periodic active noise control and
equalization, IEEE Transactions on Speech and Audio Processing, 5(4), 348358, (1997).
4. Q. Shen and A. Spanias, Frequency-domain adaptive algorithms for multi-channel active
control, Recent Advances in the Active Control of Sound & Vibration, 755-765, (1993).
5. B. Rafaely and S. J. Elliott, A computationally efficient frequency-domain LMS algorithm
with constraints on the adaptive filter, IEEE Transactions on Signal Processing, 48(6),
1649-1655, (2000).
6. J. Duan, M. Li, T. C. Lim, M-R. Lee, W. Vanhaaften, M-T. Chen and T. Abe, Active
Control of Powertrain Noise Using a Frequency Domain Filtered-x LMS Algorithm, SAE
Noise and Vibration Conference and Exhibition, St. Charles, Illinois, Paper No. 2009-01-
2145, (2009).
7. T. Kasaka, S. J. Elliott and C. C. Boucher, A novel frequency domain filtered-x LMS
algorithm for active noise reduction, IEEE International Conference on Acoustics, Speech,
and Signal Processing, 1, 403-406, (1997).
8. S. M. Kuo, R. K. Yenduri and A. Gupta, Frequnecy-domain delayless active sound quality
control system, J. Sound and Vibr., 318, 715-724, (2008).
9. E. R. Ferrara, "Frequency-domain Adaptive Filtering, Chapter 6. in Adaptive Filters, edited
by C. F. N. Cowan and P. M. Grant, Englewood Cliffs, New Jersey, 145177, (1985).
10. G. Clark, S. K. Mitra and S. R. Parker, Block implementation of adaptive digital filters,
IEEE Transaction on Acoustics, Speech and Signal Processing, 29(3), 744-752, (1981).
11. J. J. Shynk, Frequency-domain and multirate adaptive filtering, IEEE Signal Processing
Magazine, 2(1), 1437, (1992).
12. J. Duan, M. Li, T. C. Lim, M-R. Lee, W. Vanhaaften, M-T. Cheng and T. Abe, Comparative
study of frequency domain filtered-x LMS algorithms applied to vehicle powertrain noise
control, Intern. J. of Vehicle Noise and Vibration, 5(1/2), 36-52, (2009).
13. E. Sorosiak, M. Li, T. C. Lim, J. Duan, T. Abe, M-R. Lee, M-T. Cheng and W. Vanhaaften,
An active noise control system for tuning vehicle interior response, NoiseCon08, (2008).
Fig. 1 - Conventional DLMS algorithm for active sound quality control application.
x(n)
Primary
Disturbance
+
p(n)
-
d(n)
+
-
y(n)
e(n)
FFT Buffer
IFFT
FFT CLMS
S(z)
Buffer
S
.
W(z)
W(k)
x(k)
X(k) X(k) e(n) E(k) e(k)
Fig. 2 - Proposed overlap-save implementation of DLMS algorithm.
0 2000 4000 6000 8000 10000
-1
-0.5
0
0.5
1
Time (sample)
A
m
p
l
i
t
u
d
e
(
V
)
(a)
0 2000 4000 6000 8000 10000
-1
-0.5
0
0.5
1
Time (sample)
A
m
p
l
i
t
u
d
e
(
V
)
(b)
Fig. 3 - Time history of control results by using conventional DLMS algorithm of Fig. 1. a)
Steady-state pure tone signal at 100Hz with unit amplitude; (b) Steady-state pure tone
signal at 110Hz with unit amplitude. Note that V refers to voltage as a relative measure
of response amplitude.
x(n)
Primary
Disturbance
+
p(n)
-
d(n)
+
-
y(n)
e(n)
FFT FFT CLMS
S(z)
W(k)
Buffer/
Padded
with Last
Block
S
.
Buffer/
Insert
One Zero
Block
W(z)
IFFT
Zero Last N
Point
FFT
Drop Last N
Point
IFFT
x(k) X(k) X(k)
e(n)
e(k)
E(k)
Gradient
Constraint
0 2000 4000 6000 8000 10000
-1
-0.5
0
0.5
1
Time (sample)
A
m
p
l
i
t
u
d
e
(
V
)
(a)
0 2000 4000 6000 8000 10000
-1
-0.5
0
0.5
1
Time (sample)
A
m
p
l
i
t
u
d
e
(
V
)
(b)
Fig. 4 - Time history of control results by using the modified delayless frequency domain LMS
algorithm of Fig. 2. (a) Steady-state pure tone signal at 100Hz with unit amplitude; (b)
Steady-state pure tone signal at 110Hz with unit amplitude. Note that V refers to
voltage as a relative measure of response amplitude.
0 200 400 600 800 1000
-20
-10
0
10
20
30
M
a
g
n
i
t
u
d
e
(
d
B
)
0 200 400 600 800 1000
-5
0
5
P
h
a
s
e
(
r
a
d
)
Frequency (Hz)
Fig. 5 - Magnitude and phase response of secondary path transfer function.
1000 1500 2000 2500 3000 3500
Engine speed (rpm)
A
m
p
l
i
t
u
d
e
(
d
B
A
)
10dB
Fig. 6 - Comparison of the performance of various active noise control systems applied to
engine speed ramping up case. (Keys: Black line , baseline noise response; blue
line , FLMS algorithm; green line , conventional DLMS algorithm of Fig.
1; and red line , proposed DLMS algorithm of Fig. 2).