2009-01-2145

Control of Powertrain Noise Using a Frequency Domain Filtered-x LMS Algorithm

Jie Duan, Mingfeng Li and Teik C. Lim
University of Cincinnati
Ming-Ran Lee, Wayne Vanhaaften, Ming-Te Cheng and Takeshi Abe
Ford Motor Company
Copyright 2009 SAE International
ABSTRACT
An enhanced, frequency domain filtered-x least mean
square (LMS) algorithm is proposed as the basis for an
active control system for treating powertrain noise.
There are primarily three advantages of this approach:
(i) saving of computing time especially for long
controllers filter length; (ii) more accurate estimation of
the gradient due to the sample averaging of the whole
data block; and (iii) capacity for rapid convergence when
the adaptation parameter is correctly adjusted for each
frequency bin. Unlike traditional active noise control
techniques for suppressing response, the proposed
frequency domain FXLMS algorithm is targeted at tuning
vehicle interior response in order to achieve a desirable
sound quality. The proposed control algorithm is studied
numerically by applying the analysis to treat vehicle
interior noise represented by either measured or
predicted cavity acoustic transfer functions. The
simulation results show that the proposed algorithm, by
increasing block size and utilizing proper step size, can
yield a more precise targeted reduction and at the same
time suppress unintended overshoot compared to the
traditional time domain algorithm.
INTRODUCTION
Vehicle NVH (noise, vibration and harshness)
characteristics have always been an important
consideration in automotive design and manufacturing
[1]. For many potential buyers, the perceived interior
loudness and sound quality that are known major
contributors to NVH can be important. Therefore, to
satisfy the increasing customer demand for better NVH
performance, automotive engineers are interested in
designing vehicles with more pleasing sound quality. To
address this need, an active noise control (ANC) system
is developed and tested in our analysis. Compared to
the traditional cabin noise control approaches that
focused on attaining maximum attenuation, the
proposed system is meant to reshape the sound
spectrum in the passenger compartment, so that the
modified sound meets the passengers expectation from
the viewpoints of functionality and pleasantness [2-4].
In this study, the focus is mainly on powertrain noise,
which is one of the major components of the interior
response. The most significant source of powertrain
noise is the engine. Since powertrain noise is typically
dominated by the harmonics in the low frequency range,
it is a suitable candidate for our proposed active sound
tuning approach. The concept uses an adaptive
equalization technique to alter the response amplitudes
of targeted engine orders such that the perceived overall
sound quality is improved.
The use of active noise control to treat powertrain noise
has been under consideration for a number of years.
However, there are basically none implemented in
production vehicles due to a number of technical
challenges including system response time. Recently, a
novel use of frequency domain based active noise
control algorithm is proposed. This frequency domain
algorithm has several advantages compared to
conventional time domain technique. One of advantages
is the significant saving in computational cost. Also, it
allows dynamic signal to be processed block by block,
which enables most convolutions and correlations to be
performed in frequency domain via the Fast Fourier
Transform (FFT). In addition, faster convergence can be
achieved as reported by References [4-6].

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In our investigation, the fast least mean square (FLMS)
algorithm is applied to ensure that the proposed
approach is suitable for a broad range of frequencies [7].
The problem of block delay inherent in the frequency-
based analysis is compensated by adding a second filter
that is placed in the signal path, which is similar to the
estimated secondary path transfer function. The
advantage of tuning step size for each individual
frequency is analyzed and demonstrated by controlling
two distinct harmonics. Furthermore, the overshoot can
be suppressed by increasing block size or adjusting step
size at each frequency bin. Using a computer simulation
model, the effectiveness of the proposed system is
verified.
This paper is organized as follows. Firstly, the
fundamental configuration of the proposed active
powertrain noise control system is presented. Secondly,
the theoretical equation for fast least mean square
algorithm is derived in Section 2. Thirdly, in Section 3,
how computational saving can be achieved is discussed.
Finally, in Section 4, the proposed active noise control
system is implemented numerically to tune the response
of selected orders of powertrain noise and to match the
corresponding sound spectrum with a predefined set of
desired amplitudes. The adjusted step size and larger
block size is applied to overcome the overshoot problem
in the adjacent frequency bands.
ANC CONTROLLER
FAST LEAST MEAN SQUARE ALGORITHM - The
structure of a simple frequency domain active noise
control system is illustrated in Reference [3]. This simple
frequency domain algorithm produces a circular
convolution instead of a linear one. The circular
convolution results in the transformation of a linear time-
invariant filter into a periodic time-varying one. Thus, the
output of the filter is periodically nonstationary for a
stationary input [7]. To avoid the nonstationary output
signal, the fast least mean square (FLMS) adaptive filter
proposed by Ferrara [7], which is developed to perform
strictly linear convolution using the overlap-save
method [8]. Clark [9] showed that 50% overlap is the
most efficient, and thus 2N-points FFT is recommended.
The theoretical equations are derived as shown in the
ensuing discussion.
Figure 1 shows the flowchart of the FLMS algorithm. In
this algorithm, only the last N terms of the 2N-point y(k)
is used to drive the secondary source,
'
( ) [ ( ) ( 1)] ( )
T
y k y kN y kN N last N terms of y k
(1)
where
'
( ) y k is the inverse FFT operation based on the
output of the adaptive filter that is expressed as
'
( ) { ( ) ( )} y k IFFT X k W k
(2)
Since y(k) is calculated every N sample, where k is the
block index, an unbuffer device is needed to generate
the serial signal y(n) before driving the secondary
speaker. The pseudo-error signal can be defined as
'( ) ( ) ( ) e n e n d n
(3)
where e(n) is the residual error signal sensed by the
error sensor and is the difference between the primary
disturbance PT(n) and the output of secondary sound
y(n). Also, in the above equation, d(n) is the desired
sound pressure, which is synthesized according to a
certain pre-determined vehicle interior sound quality
criteria. One block of reference signal with one previous
block, and one block of pseudo-error signal padded with
N zero signals are accumulated in the buffer separately
to generate two 2N-point signal vectors, x(k) and e(k),
( ) [ ( ) ( 1) ( ) ( 1)] x k x kN N x kN x kN x kN N
' '
'( ) [0 0 0 ( ) ( 1)] e k e kN e kN N (4)
where k is the block index. Then, the signal vector x(k)
and e(k) are transformed once every N samples by a
2N-point FFT to produce a pair of frequency domain
vectors:
( ) { ( )} X k FFT x k ,
'
'( ) { ( )} E k FFT e k
(5)
Both frequency domain vectors of the filtered-x
reference and pseudo-error signals are employed in
updating the adaptive filter coefficients by adding a term
called gradient that is calculated using the complex least
mean square algorithm. To ensure strict linear
convolution of the adaptive filter, a process called
gradient constraint has to be applied. As shown in
Figure 1, this process is grouped by three functional
blocks and enclosed by a dashed rectangular. The
gradient is computed as follows:
' '
( ) { ( ) ( )} G k first Nterms of IFFT X k E k (6)
where
'
( ) X k is the conjugate of
'
( ) X k . Hence, the
update strategy for the filter weights is
( )
0
( 1) ( )
:
0
G k
W k W k FFT








(7)
The last N value of the inverse FFT transform of gradient
are force to zero. Finally, compared to the simple
frequency domain algorithm in Reference [3], after
taking the above additional processes, the adaptive filter
shown in Figure 1 performs exact linear convolution
instead of a circular one.
CONTROL SYSTEM CONFIGURATION - The proposed
frequency domain control system is shown in Figure 2.
The FLMS controller shown includes all the algorithms
such as gradient constraint, pseudo-error augment, and
overlap for reference signal, discussed earlier. Also,
both 2N-point FFT and 2N-point IFFT are applied in this
system to avoid the nonstationary output due to the
circular convolution. It may be noted that the adaptive
filter works in the frequency domain. The term D(z) in
Figure 2 is a compensation filter for the block delay.
The powertrain disturbance signal employed in this
study is measured from an actual production vehicle.
Since the feedforward control strategy requires a clean
reference signal that is highly coherent with the primary
disturbance being targeted for tuning, a tachometer
signal is used to estimate the engine crankshaft speed.
Then, the reference signal can be generated by using a
sine wave generator based on the calculated engine
crankshaft speed [10]. This procedure enables highly
coherent events between the reference signal and
powertrain disturbance. Hence, the reference signal x(n)
can be expressed as
1
( ) sin(2 / )
N
i i s
i
x n a nf f

(8)
where a
i
is the amplitude of the i
th
order, f
i
is the
frequency of the i
th
order and the f
s
is the sampling rate.
In the conventional active noise control application, the
desired response is zero. However, in this study, for the
different purposes of enhancement, attenuation or
cancellation, the desired response d(n) can be
expressed as

N
i
s
E
i i
f nf d n d
1
) / 2 sin( ) (

(9
)
where d
i
is desired amplitude of the i
th
order,
E
i
f is the
frequency of the i
th
order, and f
s
is sampling rate. The
desired signal d(n) should include all frequencies that
are targeted for tuning in the control system.
Again in Figure 2, H(z) stands for the secondary path
transfer function that is the response from the control
speaker to the receiving microphone in the passenger
compartment. This transfer function can be estimated
from either a vehicle cabin model or experiment. Also,
( ) h z

is the estimated FIR (Finite Impulse Response)

model of the secondary path transfer function using
either online or offline identification method [11]. Since
there is a secondary path transfer function H(z) following
the adaptive filter,
( ) h z

must be placed in the weight

update path to compensate for the secondary path
effect. Otherwise, the algorithm will tend to diverge. This
is basically the well known Filtered-x LMS algorithm [11].


Figure 1. Proposed FLMS active noise control algorithm.
2N-Point
FFT
Weights update
2N-Point FFT
2N-Point
IFFT
Drop
first N
2N-Point
FFT
x X Y
y
y
e
PT
d
e
Gradient
constraint
Add N
zero at
the
beginnin
Conjugate
2N-Point IFFT
Zero last N
i t

Figure 2.Basic configuration of proposed frequency domain active noise control system.

COMPLEXITY ANALYSIS
Table 1. Multiplication ratio between the proposed
frequency domain algorithm and the traditional time
domain algorithm.
Block Size Multiplications Ratio
32 1.0110
64 0.8879
128 0.7050
256 0.4975
512 0.3142
Frequency domain algorithm has a potential to save
computational cost compared to the traditional time
domain algorithm because it allows block by block signal
processing [7]. The N-th order filter, M-th order
secondary path filter and N-block size are used in this
analysis. For traditional time domain algorithm, the total
computations of adaptive filter output and weights
update require 2N multiplications. In addition, calculation
of the filtered reference signal is M multiplication.
Therefore, to produce N output samples, total number of
multiplications are N(2N+M). In the case of the
frequency domain algorithm as proposed, the adaptive
filter requires four 2N-point FFT and two 2N-point IFFT.
Also the adaptive filter weights work in complex number
form. Thus, the total number of multiplications per block
signal is N(12log
2
(N)+8+M). The ratio of real-value
multiplications required by the proposed FLMS algorithm
to traditional time domain algorithm is
2
12log 8
2
N M
N M

(10)
From the results listed in Table 1, it is clearly that the
proposed FLMS algorithm will cost less computation
time than traditional time domain algorithm when the
block size is larger than 32, because those ratios are
less than 1. When the block size is equal to 32, the
computational costs of these two algorithms are almost
the same.
COMPUTER SIMULATION
STEP SIZE - Convergent rate is different for each
harmonic that is targeted for tuning control. For one
harmonic, the convergent rate depends on the step size
and the amplitude of frequency response function of the
secondary path. Since it is very difficult to change the
secondary path, adjusting step size is the best way to
achieve faster convergence and better performance.
The amplitude of frequency response function of
secondary path also varies with frequency. Thus, the
most suitable step size for each harmonic should be
different.
For the time domain algorithm, the step size is fixed for
all frequencies. Hence, the selected step size may be
optimal for only one harmonic, but not for another
harmonic. In fact, some of the other harmonics may
become divergent if the step size is not suitable. For
instance, in Figure 4, the dashed lines show the
simulation results when a fixed step size is used to
control the harmonics at 160Hz and 288Hz using the
time domain algorithm. The step size can not go any
larger in this case; otherwise, the harmonic at 288Hz will
diverge. It is seen that the harmonic at 160Hz (blue
dashed line) converge much slower than the harmonic at
288Hz (black dashed line).
x(n)
Speed Calculator
& Sine Wave
Generator
Tachometer
Signal
Powertrain
Noise
+
PT(n)
-
d(n)
+
-
y(n)
e(n)
Buffer
Buffer Buffer
Unbuffer IFFT
FFT
D(z)

( ) h z

FFT FLMS
W(z)
H(z)
FFT
For the frequency domain algorithm, the step size can
be adjusted for each frequency bin. In Figure 4, the solid
curves show the simulation results when the step size of
the harmonic at 160 Hz (blue solid line) is 10 times
larger than the step size of harmonic at 288Hz (black
solid line). It is seen that almost the same convergent
rates can be achieved for these two harmonics in Figure
4 and Figure 5. Even though the harmonic at 288Hz
(black dotted line) converge faster using the time domain
algorithm than that using the frequency domain
algorithm (black solid line), the total MSE (mean square
error) using time domain algorithm (red dotted line)
shows lower convergent rate than that of frequency
domain algorithm (red solid line). The reason is because
slower convergent rate of the harmonic at 160Hz using
time domain algorithm does affect the total system
performance. From the solid line, we can see that for the
first 700 samples, the frequency domain algorithm does
not show convergence. This is caused by the block
delay of the frequency domain algorithm. The amplitude
of frequency response function of the secondary path
used in this study is shown in Figure 3. It is clearly seen
that the amplitude response at 160Hz is much lower
than the amplitude response at 280Hz. This implies that,
to achieve similar convergent rate, a larger step size is
required at 160Hz.
100 120 140 160 180 200 220 240 260 280 300
40
45
50
Frequency (Hz)
M
a
g
n
i
t
u
d
e

(
d
B
)

Figure 3. Magnitude of the frequency response function
of the secondary path.
0 1000 2000 3000 4000 5000 6000 7000 8000
0
0.5
1
1.5
Samples
Two harmonics control, 160Hz and 288Hz
M
e
a
n

S
q
u
a
r
e

E
r
r
o
r



Figure 4. Mean square error (MSE) of two harmonics
control using frequency domain and time domain
algorithms. Solid line frequency domain algorithm;
dashed line --- time domain algorithm; red line or ---
total MSE of two harmonics; black line or --- MSE
of harmonic at 288Hz; blue line or --- MSE of
harmonic at 160Hz.
2200 2400 2600 2800 3000 3200 3400
0.16
0.18
0.2
0.22
0.24
0.26
0.28
0.3
0.32
Samples
Two harmonics control, 160Hz and 288Hz
M
e
a
n

S
q
u
a
r
e

E
r
r
o
r

Figure 5. Expanded view of two harmonics control
results using the frequency domain algorithm. Black
solid line MSE (mean square error) of harmonic at
288Hz; blue solid line MSE of harmonic at 160Hz.
POWERTRAIN NOISE CONTROL - The performance of
proposed active control system can be evaluated using
Matlab/Simulink. The primary powertrain disturbances
along with tachometer signal were recorded on an actual
vehicle when the engine crankshaft speed is set to about
3500 rpm. The reference signal is generated by a sine
wave generator including all frequencies targeted for
control. For demonstration purpose, the desired signal is
designed to reduce the response of the third order as
much as possible and to enhance the response of fourth
order to meet the desired amplitude. A transfer function
of the secondary path, which is from the driving signal of
the secondary speaker to the error sensor location, is
synthesized from a fast numerical model for vehicle
interior acoustics [12]. The secondary path was modeled
using a 300-tap finite impulse response filter
( ) h z

for all
simulations. The lengths of frequency domain adaptive
filter are N=128 and N=256. Thus, 256-Point FFT and
512-Point FFT are conducted in the relative simulations.
The spectral magnitude of primary disturbance signals
and pseudo-error signal are shown in Figure 6. In this
simulation, the last 8192 samples of pseudo-error signal
after the convergence is taken as steady-state signal.
The desired value of the 4
th
order is labeled by asterisk.
The black curve is the original response of powertrain
noise when the control is off. Both the blue curve and
red curve are the resultant responses when the control
is on. As shown in the plot, the reduction at the 3
rd
order
is 10dB when the block size is 128, and the reduction is
more than 16 dB when the block size is 256. This is
more reduction compared to only 10 dB reduction
applying the traditional time domain algorithm with the
same operating conditions [10]. The enhanced response
of the 4
th
order is very close to the desired response for
both simulations.
Expanded view as
shown in Figure 5.
120 140 160 180 200 220 240 260 280
Frequency (Hz)
R
e
l
a
t
i
v
e

a
m
p
l
i
t
u
d
e

(
d
B
)

Figure 6. Active noise control simulation results using
the proposed frequency domain active control system.
Black line primary powertrain noise; blue line
block size 128; red line block size 256; * desired
value.
However, the unintended overshoot is seen at
frequencies neighboring the controlled harmonics. There
are two ways to suppress the overshoot and to improve
the performance of the proposed system. Firstly, it is
seen that the overshoot of the red curve is less than that
of the blue curve in Figure 7. This is suggesting that the
overshoot can be suppressed as the block size
increases because the larger block size has smaller
frequency bin.
Secondly, one of the advantages of frequency domain
algorithm is that the step size can be adjusted
independently for each frequency bin. In Figure 7, the
black curve is still the baseline response of powertrain
noise. Both the blue curve and red curve are the sound
pressure responses after the controller is activated. The
step size of the blue curve is constant, while the step
size of the red curve is variable. As shown in Figure 7,
the overshoot is decreased significantly when the step
size at frequencies neighboring controlled response are
set to zero. This result is obtained when the block size is
128. Further refinement can be achieved using larger
block sizes, such as 256.
In traditional time domain algorithm, the amplitude of the
reference signal of each order is adjusted separately,
while tuning of the variable step size in the frequency
domain algorithm provides an alternative for the
proposed active control system. Furthermore, tuning
step size for each frequency bin according to the
secondary path transfer function can lead to faster
convergence and also avoid divergence in some cases.
120 140 160 180 200 220 240 260 280
R
e
l
a
t
i
v
e

a
m
p
l
i
t
u
d
e

(
d
B
)
Frequency (Hz)

Figure 7. Comparison between fixed step size and
variable step size for each frequency bin when step size
is 128. Black line primary powertrain noise; blue
line fixed step size; red line variable step size;
* desired value.
CONCLUSION
The proposed active control system for use to tune
powertrain noise was implemented and analyzed in
frequency domain. A case study is conducted in
MATLAB/Simulink, where the engine crank shaft speed
is constant. Simulations show that the proposed system
has better performance, such as more reduction at
targeted order, than conventional time domain algorithm.
Also, the overshoot problem can be suppressed by
using variable step size or by increasing block size.
Overall, we observed that this active control system
provides faster convergence, less computations, and
less overshoot while achieving the desired control
objectives. This feature is very desirable for use in
tuning interior sound quality, and is currently being
studied further for more variety of operating conditions.
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CONTACT
Name: Teik C. Lim, PhD, PE, Fellows (ASME, SAE)
Position: Professor and Department Head
Director, Vibro-Acoustics and Sound Quality
Research Laboratory
Director, Hypoid Gear Mesh and Dynamic
Modeling Consortium
Director, UC Simulation Center
Address: Mechanical Engineering, Univ. of Cincinnati
598 Rhodes Hall, P.O. Box 210072
Cincinnati, OH 45221, USA.
Email: teik.lim@uc.edu