A Novel, Cost-Effective Method for Loudspeakers Parameters Measurement

Under Non-Linear Conditions

M. Faifer, R. Ottoboni, S. Toscani
Dipartimento di Elettrotecnica, Politecnico di Milano
Piazza Leonardo da Vinci 32, 20133 Milano (MI), Italy
Phone: +39-02-23993727, Fax: +39-02-23993703, Email: roberto.ottoboni@polimi.it

Abstract The linear model of the dynamic low-frequency
loudspeaker is widely used nowadays, and it is well known how to
assess its parameters from simple measurements. However, this is a
small-signal model which well represents the behavior of a
loudspeaker only when slight diaphragm displacements are
considered. In many cases it is important to have a more accurate
model which takes into account the nonlinear phenomena as well,
for example during the development of a feedforward loudspeaker
linearization system. It is quite simple to obtain such a model from
the small signal one by introducing the dependence of the force
factor Bl, the suspension stiffness K
ms
and the voice coil inductance
L
e
from the diaphragm position x. More difficult is to assess these
position-dependent parameters. In this paper, novel and low cost
methods for the Bl(x) and K
ms
(x) identification are proposed. Some
experimental result will be presented, and in particular a
comparison between the large-signal, low-frequency behavior of the
loudspeaker and its model (whose parameters has been estimated
with the proposed methods) will be performed. Finally, this model
has been employed in order to implement a simple model-based
method for the loudspeaker linearization. Other experimental results
will be reported and the capability of the method to reduce the
second and third harmonic distortion at very low frequencies will be
discussed.

Keywords loudspeaker nonlinearities, loudspeaker linearization,
parameters estimation, force factor, suspension stiffness.

I. INTRODUCTION
In the field of electroacoustics, one of the most discussed
research activities is represented by the analysis of the
dynamic low-frequency loudspeaker nonlinearities [1], [2],
[3], [4], [5] and [6]. In particular the study of its large-signal
behavior is very important in order to improve the reliability
and the acoustic performances [7]. Basically, a significant
reduction of the harmonic distortion can be obtained through
two different ways. First of all, this could be achieved by
performing a careful optimization of the loudspeaker design
and materials. Obviously, the manufacturing cost could
noticeably increase. The second way consists in the
development of an active linearization system, which is a sort
of algorithm which controls the diaphragm motion law. This
system can be either closed-loop or open-loop. A closed-loop
approach requires a very accurate sensor in order to estimate
the voice coil position. Such a sensor could be quite
expensive. In an open-loop solution there is no need for a
sensor, and because of the decreases of the electronic
components cost, this way is becoming more and more
interesting. In particular, if the large-signal behavior of the
loudspeaker were known, it should be possible to synthesize a
mirror filter [8] used to pre-distort the signal and thus
compensating its nonlinearities [9]. Thus, a loudspeaker
model which considers the nonlinear phenomena is required
in order to design the mirror filter. Previous studies have
shown that at low frequencies the main causes of the dynamic
loudspeaker nonlinearities are substantially three:

the suspensions stiffness K
ms
depends on the diaphragm
excursion xx;
the force factor Bl is a function of the voice coil position
with respect to the air gap;
the value of the voice coil self-inductance L
e
is related
with its position.

So, the dynamic loudspeaker nonlinearities are related with
the dependence of these three parameters on the voice coil
position [10]. The classical small-signal model can be easily
adjusted in order to take into account these phenomena. It has
been shown that the obtained model well represents the large
signal behavior of the physical device assuming that its
parameters have been properly estimated. The evaluation of
the x-dependent parameters is quite tricky. This assessment
could be performed by feeding the loudspeaker with
appropriate signals, measuring the current draw and
employing least-square optimization techniques [11]. The
parameters could be also estimated with some spot
measurements on the physical device [12]. The drawback of
this approach is that often complex and expensive equipment
is needed.
II. FORCE FACTOR ESTIMATION
Lets suppose that x=0 is the rest position of the
diaphragm. A simple and fast way to estimate Bl(x) of a
loudspeaker comes directly from the induction law. An
electromotive force e(t) arises because of the voice coil
motion. Its expression is given by:

c(t) = Bl(x) u(t) (1)

where u(t) is the moving assembly speed:

IEEE AUTOTESTCON 2008
Salt Lake City, UT, 8-11 September 2008
978-1-4244-2226-5/08/$25.00 2008 IEEE
u(t) =
Jx(t)
Jt

(2)

Knowing the instantaneous position and speed of the voice
coil, and the instantaneous value of the induced emf e(t) it is
easy to calculate Bl(x). The suggested test setup is someway
similar to that used for the determination of the magnetization
characteristic of synchronous or DC machine: a prime motor
is required in order to impress the speed to the diaphragm,
and the instantaneous open circuit voltage at the loudspeaker
terminals should be measured. As prime motor, another
loudspeaker mounted in a face-to-face push-pull
configuration can be used. In order to reduce the effect of air
compression, it is recommended to make the chamber
between the two diaphragms as small as possible. This
annulus should also be sealed with particular attention. In this
way, at low frequencies the volume displaced by the two
loudspeakers is almost the same. The instantaneous position
of the tested loudspeaker diaphragm can be measured using a
laser distance sensor with proper requirements in terms of
bandwidth and resolution. Supposing that the force factor has
to be measured in the range x
1
x x
2
, the test is performed
by feeding the motor loudspeaker with a low-frequency sine
wave so that the diaphragm of the loudspeaker under test
moves between x
1

1
and x
2
+
2
, with
1
and
2
greater than
zero. The position and the open circuit voltage of the tested
loudspeaker should be acquired with adequate sampling rate
1/T
s
. A DFT of the two signals has to be computed, and the
high frequency components (mainly due to disturbances and
quantization noise) should be removed from the spectra.
Then, the diaphragm speed is calculated performing a
frequency-domain derivation of the previously filtered
position signal. After returning in the time domain, emf,
position and speed signals must be time-aligned in order to
compensate the different delays introduced by the
measurements devices. If these delays are unknown, it is
recommended to shift the emf signal so that the zero crossings
of the emf and speed are almost aligned. The Bl(x(nT
s
)) signal
should be calculated as the ratio (sample to sample) between
the emf and diaphragm speed. Using the x(nT
s
) signal, Bl(x)
can be estimated. Finally, an analytic expression of Bl(x)
could be obtained by performing a polynomial fitting.
The proposed method has been employed in order to
estimate Bl(x) of a commercial very low frequency
loudspeaker. The device has been tested at different
frequencies (12.5 Hz, 18.75 Hz and 25 Hz) in order to
validate the proposed method. The experimental results
confirm the goodness of the approach showing a remarkable
repeatability and a very low sensitivity to the excitation
frequency. In Fig. 1 the Bl(x) curve obtained using the
proposed method is shown.


Fig. 1 Fifth grade polynomial fitting of the force factor
obtained with the proposed method
III. SUSPENSION STIFFNESS ESTIMATION
The stiffness K
ms
(x) is quite tricky to estimate; the main
reason is that the behavior of the suspension system is fairly
complex even if the employed loudspeaker model describes it
in a very simple manner. It is possible to employ a more
complete and accurate model [1], but the assessment of the
parameters becomes even more difficult.
A possible method for the evaluation of K
ms
(x) can be
obtained from the analysis of the second-order differential
equation which describes the diaphragm motion. Indicating
with F
el
(x) the elastic force due to the suspension system,
introducing the mechanical resistance R
ms
and the total
moving mass M
ms
this equation becomes:

Bl(x)i +
1
2
JI
c
(x)
Jx
i
2
=
= H
ms
J
2
x
Jt
2
+ R
ms
Jx
Jt
+ F
cI
(x)
(3)

Lets suppose that the diaphragm is located in the
equilibrium position x=X
0
because of the force due to the
current i=I
0
flowing through the coil. The relationship
between X
0
and I
0
is given by the following expression:

Bl(X
0
)I
0
+
1
2
JI
c
(x)
Jx
_
X
0
I
0
2
= F
cI
(X
0
)
(4)

where:

F
cI
(X
0
) = K
ms
(X
0
) X
0
(5)

When small current perturbations i are considered, the
previous differential equation can be linearized around
the equilibrium point:
_Bl(X
0
) +
JI
c
(x)
Jx
_
X
0
I
0
_ oi =
= H
ms
J
2
ox
Jt
2
+ R
ms
Jox
Jt
+
+_
JF
cI
(x)
Jx
_
X
0
-
JBl(x)
Jx
_
X
0
I
0
-
1
2
J
2
I
c
(x)
Jx
2
_
X
0
I
0
2
_ ox
(6)

Usually, the first derivative of the force factor and the
second derivative of the voice coil self-inductance with
respect to x are small, and can be neglected. This leads to:

_Bl(X
0
) +
JI
c
(x)
Jx
_
X
0
I
0
_ oi =
= H
ms
J
2
ox
Jt
2
+ R
ms
Jox
Jt
+
JF
cI
(x)
Jx
_
X
0
ox
(7)

The derivative of the elastic force with respect to x
calculated in x=X
0
is the differential stiffness K
ms
(X
0
). Lets
suppose that the current perturbation is a sine wave at the
angular frequency so that it is possible to employ the
phasors analysis (phasors are written in bold). Introducing the
resonance angular frequency
s
(X
0
) and the mechanical
quality factor Q
ms
(X
0
) it can be written:

6X =
Bl(X
0
) +
JI
c
(x)
Jx
_
X
0
I
0
-_

s
(X
0
)
_
2
+ ]

s
(X
0
)
1

ms
(X
0
)
+ 1
6I
K'
ms
(X
0
)
(8)

where:

s
(X
0
) = _
K
i
ms
(X
0
)
H
ms

(9)

ms
(X
0
) =

s
(X
0
)H
ms
R
ms
(10)

The equation which gives the voltage v at the terminals is:

: = R
c
i + I
c
(x)
Ji
Jt
+ Bl(x)
Jx
Jt

(11)

At low frequencies, the voltage drop across L
e
(x) is
negligible:

: = R
c
i + Bl(x)
Jx
Jt

(12)

Linearizing the expression around the equilibrium point:

o: = R
c
oi + Bl(X
0
)
Jox
Jt

(13)

Having supposed that the perturbation is sinusoidal and
substituting (8):

6F
6I
= Z
e
(, X
0
) = R
c
+
+]
Bl(X
0
) +
JI
c
(x)
Jx
_
X
0
I
0
-_

s
(X
0
)
_
2
+ ]

s
(X
0
)
1

ms
(X
0
)
+1
Bl(X
0
)
K'
ms
(X
0
)
(14)

where Z
e
(, X
0
) is the electrical impedance when the coil is
vibrating around the equilibrium position X
0
with angular
frequency . For a given X
0
, when =
s
(X
0
) it can be shown
that the electrical impedance becomes purely resistive and
reaches a maximum.
From these observations, K
ms
(X
0
) can be assessed using a
method based on the following steps:

feed the loudspeaker with a DC current I
0
so that the
position of the diaphragm is X
0
;
superimpose an AC current at the angular frequency
measure the AC voltage at the terminals;
trace Z
e
(, X
0
) magnitude plot by sweeping the current
frequency, and find
s
(X
0
);

These operations have to be repeated with different values
of X
0
according to the excursion range where the stiffness has
to be estimated.
It is possible to calculate the total moving mass M
ms
using
the added-mass technique; knowing
s
(X
0
) and supposing
that M
ms
is constant, it is easy to estimate K
ms
(X
0
). However,
in the electro-mechanical model of the loudspeaker appears
the stiffness, not the differential stiffness. Notice that:

_
K
ms
(x) =
1
x
_K'
ms
()
x
0
J x = u
K
ms
(x) = K'
ms
(x) x = u
(15)

It is recommended to perform a grade n polynomial fitting
in order to obtain an analytic expression of the differential
stiffness K
ms
(x). At this point:

K'
ms
(x) = o

n
=0
(16)

reminding the previous expression (15):

K
ms
(x) =
o

i + 1
x

n
=0
(17)

The proposed method has been used for the K
ms
(x)
identification of the 250 mm very low frequency loudspeaker
whose force factor Bl(x) has been previously evaluated.
Adopting the added mass method a moving mass of 58.6 g
has been measured. The experimental results are shown in
Fig. 2 and Fig. 3. The curves have been obtained analyzing
the loudspeaker over the excursion range from -9 mm to 9
mm with a 1 mm step. Notice that the fitting quality of the
differential stiffness obtained with a fifth grade polynomial is
very high.


Fig. 2 Fifth grade polynomial fitting of the differential suspension stiffness
obtained employing the proposed method. The marks show the experimental
points calculated from the resonance frequency measurements.


Fig. 3 - Polynomial fitting of the suspension stiffness
obtained with the proposed method
IV. LOUDSPEAKER LINEARIZATION
In the last section of this paper, a simple loudspeaker
linearization method based on the mirror filter will be
proposed. The main assumption is that the nonlinear
phenomena are entirely located in the loudspeaker and in
particular that the acoustic load is almost linear. When a low
frequency loudspeaker is mounted in a properly designed
cabinet and the pressure levels are not very high, this
hypothesis can be considered true.
If an accurate model of the loudspeaker system were
available, it would be possible to predict the instantaneous
diaphragm position for a given input voltage v. But it would
be even possible to calculate what is the voltage signal to be
applied at the terminals so that the instantaneous diaphragm
motion law is x. This can be used to design a pre-distortion
filter inserted between the source and the power amplifier
connected to the loudspeaker.
In an ideal, perfectly linear transducer, the inductance, the
force factor and the suspension stiffness are constant. The
following expression can be written:

: =
R
c
H
md
Bl
0
J
2
x
Jt
2
+ _
R
c
R
ms
Bl
0
+ Bl
0
]
Jx
Jt
+
+
R
c
K
ms0
Bl
0
x +
R
c
Bl
0
I
-1
|sZ
mu
(s)X(s)] +I
c0
Ji
Jt
(18)

where M
md
is the moving assembly mass, Z
ma
(s) is the
mechanical impedance due to the acoustic load, X(s)
represents the Laplace transform of the diaphragm position x,
while L
-1
denotes the inverse Laplace transform operator.
Now, lets consider the loudspeaker nonlinearities and
impose that the instantaneous diaphragm position x is the
same as that calculated employing the linear model with the
input voltage v. In this case, the input voltage v
f
to be applied
at the terminals results:

:
]
=
R
c
H
md
Bl(x)
J
2
x
Jt
2
+ _
R
c
R
ms
Bl(x)
+ Bl(x)_
Jx
Jt
+
+
R
c
K
ms
(x)
Bl(x)
x +
R
c
Bl(x)
I
-1
|sZ
mu
(s)X(s)] +
-
R
c
2Bl(x)
JI
c
(x)
Jx
i
]
2
+ I
c
(x)
Ji
]
Jt
(19)

where i
f
is the current circulating in the coil of the nonlinear
loudspeaker. Subtracting (18) from (19) yields:

:
]
= : +R
c
H
md
_
1
Bl(x)
-
1
Bl
0
_
J
2
x
Jt
2
+
+_R
c
R
ms
_
1
Bl(x)
-
1
Bl
0
_ + Bl(x) - Bl
0
_
Jx
Jt
+
+R
c
_
K
ms
(x)
Bl(x)
-
K
ms0
Bl
0
_ x +
+R
c
_
1
Bl(x)
-
1
Bl
0
_ I
-1
|sZ
mu
(s)X(s)] +
-
R
c
2Bl(x)
JI
c
(x)
Jx
i
]
2
+I
c
(x)
Ji
]
Jt
-I
c0
Ji
Jt
(20)

The previous expression shows how the filter distorts the
voltage v so that the nonlinear speaker behaves like a linear
one. It contains the instantaneous diaphragm position and the
current of the ideal transducer when an input voltage v is
applied at its terminals. These quantities can be easily
computed through simple linear filtering operations
employing the small signal loudspeaker model. The
estimation of the current i
f
is much more onerous since the
integration of the nonlinear model is required. However,
nowadays in many speaker is installed a shorting ring over
the pole piece in order to reduce the voice coil self-
inductance (and its dependence on x) dramatically. So, if the
voice coil inductance is very low, it is possible to neglect the
difference between the inductive voltage drops in the
nonlinear and in the linear model:

I
c
(x)
Ji
]
Jt
- I
c0
Ji
Jt
u
(21)

In addition, it is possible to neglect the effect of the
reluctance force. This assumption is true with good accuracy
at the low frequencies: the excursion are large, thus the
nonlinearities due to the suspension stiffness and the force
factor are relatively strong. So, the relationship between v and
v
f
becomes much simpler:

:
]
= : +R
c
H
md
_
1
Bl(x)
-
1
Bl
0
_
J
2
x
Jt
2
+
+_R
c
R
ms
_
1
Bl(x)
-
1
Bl
0
_ + Bl(x) - Bl
0
_
Jx
Jt
+
+R
c
_
K
ms
(x)
Bl(x)
-
K
ms0
Bl
0
_ x +
+R
c
_
1
Bl(x)
-
1
Bl
0
_ I
-1
|sZ
mu
(s)X(s)]
(22)

The estimation of the current i
f
is no longer needed, so the
computational load is greatly reduced. It is possible to
implement the pre-distortion filter on a moderate powered
DSP. However, it should be noticed that because of the
loudspeaker asymmetry the linearization method could
require the injection of a DC voltage component at the
terminals in order to center the diaphragm motion around
x=0. Since the audio amplifiers are AC-coupled, the
employment of the pre-distortion filter requires a special
power amplifier. It is important to stress that with this method
the possible presence of a DC voltage component is crucial.
The presented linearization system has been tested on the
commercial 250 mm very low frequency loudspeaker whose
K
ms
(x) and Bl(x) have been previously assessed. The
loudspeaker has been installed in a 40 l closed box, lined with
a 25 mm thick layer of wood wool. For a closed enclosure,
the mechanical impedance Z
ma
(s) can be written as follows:

Z
ma
(s) = R
mb
+ sK
mb
+s
2
H
mb
(23)

R
mb
represents the box losses, K
mb
the spring effect of the
enclosure air volume, M
mb
the mechanical mass due to the
diaphragm radiation. The voice coil self inductance is pretty
low because of a copper shorting ring near the pole piece. For
the reason reported above, the effect of the inductance can be
neglected. Under this hypothesis, the model of the complete
loudspeaker system becomes:


: =
R
c
H'
ms
Bl(x)
J
2
x
Jt
2
+ _
R
c
R
mt
Bl(x)
+Bl(x)_
Jx
Jt
+
+
R
c
K
mt
(x)
Bl(x)
x
(24)

having defined the parameters:

H'
ms
= H
md
+ H
mb
(25)

R
mt
= R
ms
+ R
mb
(26)

K
mt
(x) = K
ms
(x) + K
mb
(27)

The values M
ms
=59.1 g, R
mt
=2.35 kg/s and
K
mt
(0)=3.78N/mm have been easily measured employing the
well known small signal techniques.
As said before, the linearization system requires an
accurate model of the loudspeaker system. In order to verify
this, the loudspeaker system has been fed with sinusoidal
signals at different frequencies and amplitudes. The voice
coil current, the diaphragm position and the near-field sound
pressure signals have been recorded with a daq USB board
connected to a PC. For the current measurement a shunt and
an isolation amplifier have been employed. A laser distance
sensor has been used as position transducer, while an high
quality microphone provides the sound pressure
measurement. The voltage has been applied for a properly
short time, in order to minimize the power compression
effects. The obtained results have been compared to that
achieved by a simulation performed on the loudspeaker
system model. As an example, some results are shown in Fig.
4and Fig. 5.


Fig. 4 Measured (dotted line) and predicted (continuous line)
diaphragm position with 9V RMS 15 Hz sinusoidal input voltage
The results shows that the loudspeaker system model is
able to predict with remarkable accuracy the behavior of the
physical device even when the voice coil displacement is
high. This means that the estimation of the parameters (in
particular K
ms
(x) and Bl(x)) is quite good.


Fig. 5 - Measured (square markers) and predicted (circular markers)
on axis sound pressure spectrum at 1 m
with 9V RMS 15 Hz sinusoidal input voltage
Since a consistent model of the loudspeaker system is
available, it is interesting to test the performance of the
linearization system. In particular, its capability to reduce the
second and third harmonic distortion for a given fundamental
sound pressure level has been investigated. For each test, a
sinusoidal input voltage at the frequency f and RMS
amplitude V has been directly applied to the loudspeaker
(without pre-distortion). The on axis sound pressure at 1 m
has been recorded and in particular its fundamental amplitude
has been measured. Using the linear model of the
loudspeaker, it is possible to predict what is the sinusoidal
input voltage which gives the same fundamental SPL (sound
pressure level) assuming a perfectly linear loudspeaker. This
input signal has been processed with a MatLab program
which implements the nonlinear filter and calculates the pre-
distorted voltage. Then, this voltage has been applied to the
loudspeaker system using a DC-coupled amplifier and the
sound pressure has been acquired. Finally, to evaluate the
performance of the linearization system the sound pressure
spectrums obtained with and without the pre-distortion filter
have been compared. The tests have been performed
considering different fundamental sound pressure levels and
frequencies. Some results are shown in Fig. 6 and
summarized in TABLE I.


Fig. 6 Measured sound pressure spectrum with (circular markers) and
without (square markers) pre-distortion filter
for a 84.4 dB 15 Hz fundamental near-field SPL
TABLE I - second and third harmonic reduction
achieved with the pre-distortion filter
f [Hz]
Fundamental
[dB SPL]
Harmonic reduction
2
nd
harmonic
[dB SPL]
3
rd
harmonic
[dB SPL]
10 75.4 0.4 0.4
10 77.1 1.7 1.7
15 82.3 3.4 3.4
15 84.4 4.6 4.6
30 93.7 6.3 6.3
30 96.2 5.5 5.5
45 99.4 5.8 5.8

It can be noticed that at very low frequencies, where the
effect of the voice coil inductance can be neglected and the
voice coil excursions are large, the method can achieve a
significant reduction of the second and third harmonic
distortion. At the higher frequencies the method becomes less
effective because the nonlinear effects of the inductance get
stronger while those related to the stiffness get weaker. The
assumptions on which the method is based on becomes not
completely true.

V. CONCLUSION
Two methods for the assessment of Bl(x) and K
ms
(x) has
been proposed. These methods are very simple and do not
require expensive instrumentation; in addition, they can be
easily automated. The suspension stiffness and force factor of
a commercial loudspeaker have been evaluated using the
proposed procedures. This has permitted to build up a
nonlinear model of this loudspeaker mounted into a closed
enclosure.
Tests shown how the model can accurately predict the non
linear low frequency behavior of the physical devices. The
obtained results prove the goodness of the model parameter
estimation.
Finally a model based sensorless loudspeaker linearization
system has been proposed. Its capability to reduce the second
and the third harmonic distortion has been proved.

REFERENCES

[1] M.H. Knudsen, J.G. Jensen, Low-Frequency Loudspeaker Models that
Include Suspension Creep, Journal of the Audio Engineering Society,
Volume 41, pp. 3-18, January/February 1993.
[2] M. Navarri, E. Bellati, F. Tordini, R. Toppi, A Novel Synthesis
Approach to Loudspeaker Design, presented at the Italian Audio
Engineering Society Seminar on Loudspeaker and Nonlinearities,
Milan, September 20 2003.
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