Proceedings of the 2007 IEEE

International Conference on Mechatronics and Automation

August 5 - 8, 2007, Harbin, China

An Approximate Realization Method of the Square Root

Signal Processing Algorithm of the Audio Directional
Loudspeaker
Chen Min1, Huang Da-gui2, Xu Shou-heng3, Xu Li-mei1
1. Institute of Astronautics & Aeronautics, University of Electronic Science and Technology of China, Chengdu, Sichuan
Province, China
2. School of Mechatronics Engineering, University of Electronic Science and Technology of China, Chengdu, Sichuan
Province, China
3. Department of Technique, Xichang Satellite Launch Center, Xichang, Sichuan Province, China
chm_china@163.com, dg_huang@vip.163.com, xulimei@uestc.edu.cn
Abstract - 7he realization of the square root signal processing
algorithm is investigated for the audio directional loudspeaker.
First, the fundamental principle of the audio directional
loudspeaker is briefly introduced. Then, an nth-order
approximation algorithm is proposed to realize the square root
signal processing algorithm which can not be realized physically
for its infinite signal bandwidth requirement. It is shown that the
requirement of the infinite signal bandwidth of the square root
algorithm is loosed successfully by utilizing the nth-order
approximation algorithm which can be achieved by the physical
system easily. It is also verified by experiments that the nth-order
approximation methods can provide the lower total harmonic
distortion (THD) for the square root algorithm, but it does not
imply that a higher-order approximation algorithm will get lower
THD than a lower-order approximation algorithm.

useful and effective signal processing method, because it will

not introduce harmonic distortion into the processed signal
[3,4,10]. However, an infinite signal bandwidth is required by
the square root algorithm, which can not be realized by a
physical system[3,4,10]. An approximate realization method
of the square root method will be investigated in this paper to
solve the infinite signal bandwidth requirement problem.
II. FUNDAMENTAL PRINCIPLE OF AUDIO DIRECTIONAL LOUDSPEAKER
A. Parametric Array
A physics professor of Brown University, Peter Westervelt
proposed the concept of parametric array firstly in 1963 [11].
Fig.1 is a schematic diagram of parametric array. An intense
ultrasonic modulated by an audible sound is emitted into air
by the transducer. Along the propagation path, audible sound
will be demodulated gradually from the ultrasonic by the
nonlinear propagation effect of the ultrasonic in air, by which
a continuous end-fire virtual source array is constructed. This
virtual source array is called parametric array, which focuses
most of the energy of the audio signal in a narrow beam.
Therefore, the audible sound is hardly heard outside the audio
beam. The fundamental principle of audio directional
loudspeaker is properly structured on the parametric array.

Index Terms - audio directional loudspeaker, square root signal

processing algorithm, nth-order approximation algorithm, signal
bandwidth, total harmonic distortion

I.

INTRODUCTION

Audio directional loudspeaker is a novel loudspeaker which

can produce audible sound with high directionality by utilizing
the nonlinear propagation effect of ultrasonic in air, which
achieves the distribution of sound in a narrow beam zone
firstly. As a revolutionary loudspeaker, audio directional
loudspeaker now is also known as audio beam loudspeaker[1],
audio spotlight[2,3], hypersonic sound system[4], parametric
loudspeaker[5,6,7] and parametric array[8,9,10] in literatures
since its uniform name is not yet formed. For its ability to
produce audible sound with high directionality, it will be
called audio directional loudspeaker in this paper. Among the
key technologies of audio directional loudspeaker, signal
processing method plays a very important role. With the
development of audio directional loudspeaker, many signal
processing methods, such as the double side band(DSB)[2,4,5]
algorithm, single side band(SSB) algorithm[4], truncated DSB
algorithm[4], square root algorithm[3,4,5], and so on, are
proposed. Among them, the square root algorithm is a very

1-4244-0828-8/07/$20.00 2007 IEEE.

Fig.1 Parametric array

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B. Berktay Far-Field Solution

A more accurate parametric array theory is proposed by
H.O. Berktay in 1965[12]. Almost all the algorithms used in
current signal process for audio directional loudspeaker are
deduced according to H.O. Berktays parametric array theory.
Berktay assumes the primary wave has the form
P (t ) P E (t ) sin(Z t ) .
(1)
where P0 is the pressure amplitude of the primary wave,
Zc is the carrier wave frequency, and E (t ) is the envelope
function. For a point along the transducer radiation axis, the
secondary (or self-demodulated) signal is
P02 A
wE 2
P2 (t )
E (W ) .
(2)
16SU0 c04 zD wt 2
1

The output audible sound of the square root algorithm has

only fundamental frequency, and has no harmonic, which
implies that there is no harmonic distortion in the square root
algorithm. It is shown that the square root algorithm is far
more outstanding than the DSB algorithm.
The spectrum of P (t ) in the square root algorithm is
shown in Fig.2, where the input signal is a standard sinusoidal
wave of 2kHz, the frequency of carrier wave is 40kHz, the
sampling frequency is 120kHz, and the modulation factor m
is 1. The frequency content is a continuous function, and
distributes over the whole frequency axis even if the sampling
frequency rises to a very high level, which means an infinite
signal bandwidth is required. However, it is unrealistic for a
physical system to provide an infinite signal bandwidth, and it
is necessary to find a physical system with a finite signal
bandwidth to realize the square root algorithm.
1

where A is the cross-sectional area of primary wave, E is

the nonlinear coefficient of primary wave, U 0 is the ambient
air density, c0 is the propagation speed of sound in air, z is
the distance along the beam, D is attenuation factor of the
primary signal, t is time, and W t  z / c0 is the retarded
time.
A result can be deduced from (2): the self-demodulated
audible signal of parametric array is proportional to the second
time-derivative of the envelope squared.
w2
P2 (t ) v 2 E 2 (t )
(3)
wt
Although (3) is called Berktays Far-Field Solution, it is
valid for near-field too, and it can account for how the audible
signal is demodulated from the ultrasonic.

B. Approximate Algorithm of Square Root Algorithm

Because most of the frequency content with relative higher
level is within a finite bandwidth, a finite signal bandwidth
system can by obtained by the approximation of the square
root algorithm. Fortunately, (6) can be expanded into
Maclaurins series when mf (t ) [1,1]
P (t )
1

According to (8), a series of approximate algorithms with

different precision can be obtained. The first-order
approximation algorithm, the second-order approximation
algorithm and the third-order approximation algorithm can be
expressed respectively as follows:

P1 (t )

P [1 

1
mf (t )] cos(Z t ) ,
2

P (t )

P {1 

1
1
mf (t ) 
[mf (t )] } cos(Z t ) ,
2
24

(9a)

P0 1  mf (t ) cos(Z c t ) .

(6)

The output audio signal of audio directional loudspeaker

can be deduced by (3)
P (t ) v  P mZ cos Zt .
(7)
2

P (t )

A. Theoretical Square Root Algorithm

If a DSB algorithm is applied in the signal process of audio
directional loudspeaker, the P (t ) will be expressed as follow
P (t ) P (1  mf (t )) cos Z t .
(4)
where m is the modulation factor, and f (t ) is the input
audio signal of the audio directional loudspeaker. A generic
cosZt can be used to substitute for f (t ) here to analyze the
DSB algorithm, where Z is the frequency of the input audio
signal. Then, the output audible signal of the audio directional
loudspeaker can be obtained according to (3)
P (t ) v  P (2mZ sin Zt  2m Z cos 2Zt ) .
(5)
As shown in (5), a 2nd harmonic appears, and its amplitude
equals that of fundamental wave, which implies a high
harmonic distortion exists in the DSB algorithm. Therefore,
the square root algorithm is presented to overcome the
disadvantage of the DSB algorithm, and it can be expressed as
following form
0

III. SQUARE ROOT ALGORITHM

1 3
1
1
mf (t ) 
[mf (t )]
[mf (t )] 
246
24
2
. (8)
1 3 5

[mf (t )]  } cos(Z t )
2 4 6 8
P {1 

Fig.2 Spectrum of the square root algorithm

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(9b)

P (t )
1

P {1 
0

1
1
mf (t ) 
[mf (t )]
2
24

1 3

[mf (t )] } cos(Z t )
246

(9c)

The precision of these algorithms can be analyzed in time

domain roughly. When the input signal f (t ) is a standard
sinusoidal wave of 2kHz, and the modulation factor m
equals 0.8, the modulation envelope E (t ) and the original
wave are shown in Fig.3. When the order of approximation
algorithm increases, the corresponding approximation curve
comes closer to the original signal curve, which implies the
precision
of
the
approximation
algorithm
rises
correspondingly. Accompanying with the increase of the order,
the required signal bandwidth rises. The band of the input
signal f (t ) is 20Hz~20kHz, and the signal band of the
first-order approximation algorithm, the second-order
approximation algorithm and the third-order approximation
will be 20~60kHz, 0~80Khz, and 0~100kHz respectively
according to (9a), (9b), and (9c). Therefore, the higher order
results in the higher required signal bandwidth and the
realization difficulty of actual physical system. The order of
approximation algorithm should be controlled within a limited
range.
An effective factor related closely to the precision is the
error between the original wave curve and the curve produced
by a certain approximation algorithm. The relative error of the
first-order approximate algorithm, second-order approximate
algorithm and third-order approximation algorithm are shown
in Fig.4. When the modulation factor m equals 0.5, the
maximal relative error of the first-order approximation
algorithm is about 6.1%, while that of the second-order
approximation algorithm is about 1.65%, and that of the
third-order approximation algorithm is about 0.55%.
Therefore, the higher the order is, the lower the error will be,
and it is possible that higher precision will be obtained
through providing a higher-order approximation algorithm.

Fig.3 Original wave vs. approximate wave

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Fig.4 Relative error between approximate algorithms and original wave

Fig.5 Relation between m and relative error

The relative error is also related to the modulation factor

m . As shown in Fig.5, with the increase of the modulation
factor m , the maximal relative error rises correspondingly.
When m is a certain value, the relative error of the
first-order approximation algorithm is lager than that of the
second-order approximation algorithm, and the relative error
of the second-order approximation algorithm is larger than
that of the third-order approximation algorithm. If the
tolerance relative error in time domain is about 2%, the
second-order approximation algorithm and third-order
approximation algorithm can both satisfy the requirement
when the modulation factor is 0.5, but even the third-order
approximation algorithm can not satisfy the requirement when
the modulation factor is bigger than 0.7. Then, a higher order
seems to be required, which implies that a higher signal
bandwidth is required and it is more difficult to realize the
algorithm by physical system.

self-demodulated signal when it is sampled by computer and

transformed into voltage, which is proportional to its sound
pressure amplitude. As shown in Fig.6(b), the
self-demodulated signal is mainly composed of the
fundamental frequency, second-harmonic and higher-order
harmonics, and the total harmonic distortion (THD) is mostly
introduced by the second-harmonic. The measured THD of the
square root algorithm is about 32.43%.
The self-demodulated signal of the first-order approximate
algorithm is shown in Fig.7. Its distortion in time domain is
higher than that of the square root algorithm. It is mainly
composed of the fundamental frequency and second-harmonic,
which implies that the main distortion source is the
second-harmonic. Its THD is higher than that of the square
root algorithm, and the measured THD is about 44.89%,
which means that it cant reduce the THD for the square root
algorithm.

IV. EXPERIMENTS
Some experiments have been done to verify the
improvement effect for the sound fidelity of the nth order
approximate square root algorithms. The same configuration
of these experiments is as follows: (1) The frequency response
range of the microphone is 20-16200Hz. (2) The planar
transducer size is 300mm300mm. (3) The distance between
the microphone and the transducer is 2m. (4) The input signal
is a standard sinusoidal signal of 2kHz. (5) The modulation
factor m is 0.7. (6) The frequency of the carrier wave is
40kHz. (7) Aiglent 35670A dynamic signal analyzer is
adopted as the testing instrument.
As shown in Fig.6(a), the self-demodulated signal of the
square root algorithm is an approximate sinusoidal signal of
2kHz. Compared to the input standard sinusoidal signal of
2kHz, an apparent distortion exists in the wave crest. The
vertical axis of Fig.6(a) denotes the voltage amplitude of the


(a) Time domain

(a) Time domain

(b) Frequency domain

(b) Frequency domain

Fig.6 Self-demodulated audio signal of the square root algorithm

Fig.7 Self-demodulated audio signal of the first-order approximate algorithm

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As shown in Fig.8, the waveform distortion of the

self-demodulated signal produced by the second-order
approximate algorithm in time domain is lower than those of
the square root algorithm and the first-order approximate
algorithm. In frequency domain, the fundamental frequency,
second-harmonic and triple-harmonic are the main
components of the self-demodulated signal. The amplitude of
the triple-harmonic is very close to that of the
second-harmonic, which implies the harmonic distortion is
mostly caused by both the second-harmonic and the
triple-harmonic, and the measured THD is about 17.12%.
The self-demodulated signal of the third-order approximate
algorithm is shown in Fig.9. An unexpected waveform
distortion appears in time domain, which is higher than that of
the second-order approximate algorithm. The fundamental
frequency, second-harmonic, triple-harmonic and the
fourth-harmonic mostly result in the harmonic distortion, and
the fourth-harmonic increases notablely, which is below 40dB

(a) Time domain

(a) Time domain

(b) Frequency domain

Fig.9 Self-demodulated audio signal of the 3rd-order approximate algorithm

in the other three algorithm. As far as the THD of the

third-order approximate algorithm is concerned, the measured
THD is about 23.07%.
In brief, the first-order approximate algorithm provides a
coarse approximation for the square root algorithm, but it
cant reduce the THD of the square root algorithm. The
second-order and triple-order approximate algorithm can both
reduce the THD of square root algorithm dramatically.
However, it does not imply that the higher the order of the
algorithm is, the smaller the THD will be. Moreover, another
conclusion will be drawn from the results of the experiments
that the waveform distortion in time domain mostly appears
when the sound pressure is high.

(b) Frequency domain

Fig.8 Self-demodulated audio signal of the 2nd-order approximate algorithm

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V. CONCLUSIONS
It is feasible to reduce the requirement of the infinite signal
bandwidth requirement of the square root algorithm by
utilizing the nth order approximation algorithm, which can be
realized by physical system easily. However, when the
modulation factor rises to a high level, the precision of the
nth-order approximation recedes. Therefore, the nth-order
approximation algorithm is reasonable only when the
modulation factor is low. Although the nth-order
approximation algorithms can reduce the THD of the square
root algorithm, it does not imply that a higher-order
approximation algorithm will produce smaller THD than a
lower-order approximation algorithm. Furthermore, the
distortion of waveform mostly appears when the sound
pressure is high.
ACKNOWLEDGMENT
The work was supported by the national nature science
foundation of China (No. 60302001). The authors wish to
thank the national nature science foundation committee of
China for making the audio directional loudspeaker research
work go smoothly and all colleagues in University of
Electronic Science and Technology of China for providing
helps to the project.
REFERENCES
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Olszewski, Dirk; Prasetyo, Fransiskus; Linhard, Klaus, Steerable highly

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