The 21st International Congress on Sound and Vibration

13-17 July, 2014, Beijing/China

PARAMETRIC LOUDSPEAKER: FROM THEORY TO APPLICATIONS

Jun Yang, Jing Tian
Institute of Acoustics, Chinese Academy of Sciences, Beijing, China 100190
e-mail:jyang@mail.ioa.ac.cn

Woon Seng Gan

School of Electrical and Electronic Engineering, Nanyang Technological University, Singapore 639798
A review of the current state of parametric loudspeaker from the aspects of modelling, measurement, signal processing, implementation and applications is presented in this paper. A variety of well-established and emerging numerical methods to predict the propagation of finite
amplitude sound beams are discussed and compared. The accurate measurement of the difference-frequency sound is another research interest related to parametric loudspeaker. The
measurement methods using an acoustic filter are outlined, and the approaches by taking advantage of condenser microphone sensitivity characteristics or phase cancelling technique are
also shown. The signal processing algorithm to reduce the harmful effects of the harmonic
distortion also has constantly attracted the attention of researchers. The processing method
based on Berktays solution together with the new approaches using psychoacoustic phenomenon or Volterra filter modelling are given. Furthermore, the detailed implementation of parametric loudspeaker is summarized and its potential application is listed. This paper will focus on the recent advance in the parametric loudspeaker field from theoretical perspective as
well as engineering perspective to promote the theoretical and practical development of the
parametric loudspeaker.

1.

Introduction

By utilizing the nonliearity of the acoustic waves in air, parametric loudspeaker can be used to
create sound source with higher directivity at a significantly smaller aperture compared to conventional loudspeakers. The parametric loudspeaker has been used in underwater sonar applications for
decade, here it is generally referred as the parametric array deployed in air 1-4. The basic idea of the
parametric array was originally conceived by Westervelt about 50 years ago based on the scattering
of sound by sound 5-6. The original work by Westervelt 7 addressed the generation of primary waves,
namely difference-frequency signal from two high-frequency collimated beams. The directivity of
the parametric array is attributed to an end-fire array of virtual sources, which are the by-products
of the nonlinear interaction between the primary waves propogating in the medium. A more complete explanation of the parametric array was given by Berktay 8 and his analysis was not limited to

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21st International Congress on Sound and Vibration (ICSV21), Beijing, China, 13-17 July 2014
two primary waves. Assumed that the primary wave is in the form of p1 P0 E (t )sin ct with concept
of the modulation envelope, a farfield solution can be obtained
(1)
p2 P02 2 E 2 (t ) / t 2 ,
where t is the time, E(t) is the envelope function, P0 is the initial sound pressure level (SPL) of the
carrier and p2 is the demodulated signal. The Berktays farfield solution, which defines the demodulated secondary waveform generated along the axis direction of the beam, is proportional to the second-time derivative of the square of the envelope. Parametric loudspeakers have attracted intensive
interest since the theoretical study of underwater application was initiated in 1950s. Since the discovery of parametric array by Westervelt, hundreds of papers and technical reports on this topic
have been published from theoretical and experimental points of view. However, almost all the papers before 1975 focused on exploratory examinations and applications in underwater. An experimental investigation, conducted by Bennett and Blackstock 1, has proved that an audio sound with
high directivity can be generated only in the target region without affecting other areas when two
primary waves of different frequencies interact in air. Yoneyama et al. have modulated ultrasonic
waves with audio signals using double sideband amplitude modulation (DSBAM) method, which
was regarded as the beginning of the practical application of parametric loudspeakers in audio
products 2.
The parametric loudspeaker has attracted considerable interest in the acoustic engineering
field, and a substantial number of academic research papers have been published about the basic
theory and key technologies. Some important advances have been achieved both in the engineering
and research fields of the parametric loudspeaker. In this paper, we review some of these key research works, and report on new advances in the parametric loudspeaker, including numerical modelling for nonlinear acoustics, difference-frequency sound measurement, signal processing techniques, implementation issues and applications.

2.

Numerical modeling for nonlinear acoustics

It is well known that the Khokhlov-Zabolotskaya-Kuznetsov (KZK) equation can accurately

describe the propagation of finite amplitude sound beams by combining the effects of diffraction,
absorption, and nonlinearity under a parabolic approximation 9. The KZK equation is widely used
for theoretical prediction of the nonlinear field of a directive sound beam, and the validity of the
KZK equation for the parametric array has been verified by many researchers 10-11. This model
equation is described as 12:
2 p c0 2
3 p
2 p2
p 3 3
,
(2)
zt 2
2c0 t 2 0c03 t 2
where p is the sound pressure, z is the coordinate along the axis of beams, c0 is the sound speed,
0 is the medium density, is the sound diusivity related to thermoviscous absorption, and is
the nonlinearity coecient. Moreover, 2 2 / x 2 2 / y 2 is a Laplacian that operates in the
x y plane perpendicular to the axis of the beam ( z axis), and t t z / c0 is the retarded time.
The three terms on the right-hand side of Eq. (2) account for diffraction, nonlinearity, and
thermoviscous absorption respectively. Khokhlov and Zabolotskaya derived the lossless form of the
KZK equation, i.e., KZ equation 13, and Kuznetsov considered the effect of thermoviscous absorption 14. There is still no explicit analytical solution to the complete KZK equation. Instead, numerical methods have been developed, including frequency-domain algorithm 15-19, time-domain algorithm 20-21 and combined time-frequency domain algorithm 22-26, and a review of these methods can
be found in the reference 9. Bakhvalov and his coworkers have carried out the numerical study
about the KZK equation in a series of articles 22-26. They compute the effects of absorption and

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diffraction in the frequency domain, and effect of nonlinearity in the time domain. The first spectral
algorithm was Bergen Code which was developed by Aanonsen et al. to calculate the near field of
finite-amplitude sound beams 15. The core of Bergen Code is to expand the sound signals into the
Fourier series and solve every frequency component individually. Subsequently, the spectral algorithm was expended to calculate the far field 16, bi-frequency source 17, focusing beams 18 and nonaxisymmetric source 19. The time-domain algorithm was developed by Lee and Hamilton 20-21,
known as the Texas Code, which avoids the truncation of the Fourier series in the spectral algorithm.
The advantages of using the time-domain KZK algorithm are 1) its ability to compute arbitrary time
waveform, and 2) computational efficiency 27 as compared to other previous methods. However, the
original Texas Code just aimed at the thermoviscous medium and did not consider the relaxation
effect, which was completed by Cleveland et al. 28. Liu et al. developed a three-dimensional timedomain finite difference algorithm for solving an augmented KZK equation 29 based on Transformed beam equation (TBE), where the relaxation effect is also included. Figure 1 illustrates the
sound fields of the difference frequency with the same parameters in the reference 19, where Figure
1(a) is the axial fields and Figure 1(b) corresponds to the beam patterns at z 6 m in the y direction. The results calculated by Kamakura et al. 19 are given in Figure 1 for comparison. One can see
that our results agree well with those obtained by Kamakura et al.19. The maximum deviation is less
than 1 dB and exists in the situation of high sound pressure level ( p0 128 dB).

Figure 1. The sound fields of the difference frequency calculated by using the present 3D time-domain
code: (a) the axial field; (b) y-directional beam patterns at z=6 m.

In the case of weak nonlinearity, the quasilinear solutions to the KZK equation have been obtained for the fundamental and second-order sound fields using the successive approximation
method 9, which are based on the assumption that the second order solutions are small corrections to
the corresponding linear theory. It ignores all the harmonic components larger than second order.
Under the quasilinear approximation, the sound fields are reduced to the sum of the first order and
second order fields which can be expressed as the integral. This is a heavy numerical task. To solve
this problem, the Gaussian beam expansion technique has been widely used in modelling the fundamental and the second-order wave field for its high computational efficiency 30-38. The principle
of this technique is to decompose the source distribution function into the superposition of a small
number of Gaussian functions. An advantage of using the Gaussian-beam expansion technique is
that an analytical description of the fundamental sound field propagation can be obtained, while a
simplified algorithm for the second-order sound fields can be developed, leading to fast computation. Ding 30 used this approach to simulate elliptical and rectangular transducers. Yang et al. 35-36
obtained the expansion coefficients for rectangular and elliptical planar piston transducers in a twodimensional spatial Fourier transform (k-space) domain. An overview of the Gaussian-beam expansion technique has been given in the reference 39. The key challenge here lies in the determination
of the coefficients of Gaussian functions. In the previous studies, Wen and Breazeale developed the
spatial domain method 37-38 and Sha et al. adopted the k-space domain method. However, the methICSV21, Beijing, China, 13-17 July 2014

21st International Congress on Sound and Vibration (ICSV21), Beijing, China, 13-17 July 2014
ods which use nonlinear optimization algorithms are complicated and time-consuming 35-36. This
brings forth the need to find a simple algorithm to calculate the Gaussian beam expansion coefficients. Ding and Zhang proposed a linear scheme in spatial domain 33. The obvious advantage of
this scheme is the simple determination of the Gaussian beam expansion coefficients by solving a
set of linear equations. However, since half of the predetermined coefficients are real, the practicality of this scheme is limited. Liu et al. developed a fast algorithm to determine the Gaussian function coefficients for a more accurate approximation with two-stage procedures 40. Firstly, two real
coefficients are estimated by a simple search approach, and then the least mean square (LMS) algorithm is adopted for determining the optimal expansion coefficients. Compared to the previous
methods, the new method has a better accuracy in far field. To exhibit the good performance of this
algorithm, here a simple example of the diffraction field of a circular source with 15 term Gaussian
beam expansion coefficients is given. The source distribution function is given in Figure 2(a). It can
be seen that the current result gives a better fit of the original source function than that given by
Wen and Breazeales results. Figure 2(b), 2(c), and 2(d) illustrate the axial field, the relative error of
the axial far field, and the pro-file distribution at distances of 0.25z0 (z0 is Fresnel distance), respectively. It can be seen that our results agree well with the Rayleigh integral (with ka = 107.8) except
in the very near field. Compared with Wen and Breazeales results, our results are better in far field
(the relative error in Figure 2(c) is 0.05% versus 1%).

Figure 2. Comparison of the wave field of the uniform piston source calculated by using the coefficients of
the present method, those of previous method, and Rayleigh integral: (a) the source distribution function;
(b) the axial field; (c) the relative error of the axial far field; (d) z = 0.25z0.

Since the KZK equation is derived from a full nonlinear wave equation known as the
Westervelt equation under the parabolic approximation, it introduces errors at field points that are
far away from the acoustical axis (>20) at locations close to the source 9. Several approaches have
been proposed to overcome such limitations, most of which are based on the Westervelt equation.
Huijssen et al. obtained the axial and lateral beam profiles of harmonic components by applying the
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21st International Congress on Sound and Vibration (ICSV21), Beijing, China, 13-17 July 2014
explicit finite-difference time-domain (FDTD) scheme to the Westervelt equation 41. Based on the
Neumann iterative solution, Huijssen and Verweij proposed a novel method to solve the lossless
Westervelt equation , where the nonlinear term is treated as a nonlinear contract source 42. Kamakura et al. solved the Westervelt equation by introducing the split-step Padapproach and extended
its applicability limit of propagation angle to 40, which is twice as wide as that of the KZK equation 43. Recently, Angular Spectrum Approach (ASA) has been widely used in ultrasound simulations due to its rapid computation time 44-45. ASA was firstly introduced in nonlinear acoustics by
Alais and Hennion to analyze sum- and difference frequency generation in the parametric array 46.
Landsberger and Hamilton studied the second harmonic generation of sound beams in immersed
elastic solids 47. A simulation package named as Abersim has been provided by Frijlink et al. to
solve the directional Westervelt equations in Cartesian 2D and 3D grids 48. Although it adopted
ASA in the calculation of diffraction effect, its computation time will increase dramatically if the
nonlinear term is taken into account. Du and Jensen adopted the ASA for solving the nonlinear
wave equation for the second harmonic, using the source generated by Field II based on the
Westervelt equation 49. It computes much faster than Abersim, since the ASA is an analytical solution and the simulated point can be computed in one step. Based on the Gaussian beam expansion
technique in the quasilinear approximation, Cervenka and Bednarik also obtained a formula to calculate the difference-frequency wave by a baffled axisymmetric piston with the method of successive approximations 50 .

3.

Difference-frequency sound measurement

The accurate measurement of the difference-frequency sound in the parametric loudspeaker is

vital for understanding the principle of the parametric loudspeaker and also the practical design in
audio engineering. However, spurious sound 51 will be generated as a result of nonlinearity at the
receiving system caused by the existence of the finite-amplitude ultrasonic waves, which are referred as the primary waves. The measured signal will be a combination signal including both the
difference-frequency sound from the parametric loudspeaker effect and the spurious signal. Hence,
accurately measuring the difference-frequency sound will become a problem, and the situation is
particularly worse in the nearfield and near the beam axis.
A few methods have been proposed to measure the difference-frequency sound of parametric
loudspeaker in the literature by adopting an acoustic filter mounted in front of the receiving
transducer to suppress the levels of the primary waves. In the first experimental verification on the
parametric loudspeaker in air, Bennett and Blackstock 1 adopted a dome-shaped filter made of a 2.5
mm thick clear plastic material, which reduced the SPL of the primary waves by about 20 dB at a
cost of a 3.5 dB SPL reduction for the difference-frequency signal. The effect of nonlinearity within
the receiving system was also found in water by Garrett et al. 52 and it was proportional to the
product of the primary wave at the measuring point, which was supported by the experimental
results for both the axial propagation and the beam patterns. When Lucas et al. 53 investigated the
focused parametric array in water, they implemented a sheet of rubber (1.3 cm thick) as an acoustic
filter, which provided an attenuation of 16 dB for the primary wave with an undesired 2.5 dB
reduction for the difference-frequency signal. Kamakura et al. 54 used a sheet of air-pad in their
experiments and it reduced the primary wave at 40 kHz by over 20 dB. Havelock and Brammer 55
installed a planar sound absorber made of four layers of 3.18 mm felt in their experiments and
achieved an SPL reduction at 30 kHz by over 20 dB with an unwanted attenuation for the audio
frequencies by 3-5 dB. Toda 56 presented a complex acoustic filter having four polymer layers,
which was found to have a 30-40 dB attenuation in the range of 30-40 kHz with an unwanted 5 dB
deviation in the range of 10 Hz-10 kHz. Humphrey et al. 51 adopted a 30 mm thick polyurethane
panel as the acoustic filter when they evaluated the underwater acoustic performance of panels by
using a parametric array. Wygant et al. 57 adopted a thin sheet of Saran film as the acoustic filter,
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and they found that the experimental results with the acoustic filter match well with the simulation
results. A 16-mm-diameter aluminum plate was developed as the acoustic filter by Ye et al. 58 and a
15 dB attenuation of the primary wave with an undesired 5 dB deviation in the audible range were
found by using the filter. The frequency response of the parametric loudspeaker was measured
without and with the acoustic filter as shown in Figure 3(a). It was clearly confirmed that the
spurious signal was independent of the difference frequency, in the case of without the acoustic
filter. The audible sound SPL generated by the parametric loudspeaker is proportional to f dn , where
n 1.6 . In order to further distinct the audible sound generated by the parametric loudspeaker from
the spurious signal of the transducer, the axial SPLs of the difference frequency sound was
measured and shown in Figure 3(b). The spurious signal including the transducer nonlinearity and
radiation pressure was modeled by p sn Kp a p b , where p a and pb were the peak amplitudes of the
primary waves at the transducer location, respectively. The constant number K 2.8 10 4 Pa 1 was
determined by curve fitting from the measurement results.
As can be seen from the above literature, using these proposed acoustic filters, the primary
waves at the receiving system were definitely suppressed and so did the spurious sound; however,
the measured difference-frequency sound was less affected.

Figure 3. (a)Frequency response of the parametric loudspeaker (b) Axial SPLs of the parametric loudspeaker.

Recently, two alternative methods instead of the acoustic filter in the measurement setup are
also presented. Ju et al. proposed a technique by taking advantage of the sensitivity characteristics
of the condenser microphone to reduce the levels of the primary waves with a large grazing angle 59.
But it will also affect the sensitivity in the audible range, especially when the signals frequency is
higher than 10 kHz. Based on the phase-cancellation 60 and the Gaussian beam expansion technique
30, 35, 39
, Ji et al. proposed an alternative method for measuring the difference-frequency sound accurately without using any traditional acoustic filter 61. But it is only suitable for the axial measurement.

4.

Signal Preprocessing Techniques

4.1 Processing method based on Berktays solution

Two major disadvantages of parametric loudspeakers have been noticed from practical applications, as well as from the Berktays solution: 1) the square operation of the envelope function
inevitably leads to the generation of harmonic distortions; 2) the second derivative operation behaves as a high-pass filter, which yields a poor frequency response at low frequency range, generally below 500 Hz.
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21st International Congress on Sound and Vibration (ICSV21), Beijing, China, 13-17 July 2014
By far, amplitude modulation is the dominant modulation technique used in studies and applications of parametric loudspeakers. Though the original DSBAM requires fewer computational resources, the harmonic distortion existing in the secondary wave greatly degrades the sound quality,
which can be observed from both the numerical simulations and the nonlinear Volterra modeling
results. During the past three decades, various preprocessing approaches in the modulation stage
have been proposed to alleviate the harmonic distortion. One solution introduced by Kamakura et al.
62
is to use single sideband amplitude modulation (SSBAM), which transfers the energy of two
sidebands in DSBAM to one sideband, and thus lowers the bandwidth requirement of the ultrasonic
transducer. As shown in Figure 4, SSBAM is essentially a type of quadrature modulation using two
orthogonal carriers as cos(c t ) and sin(ct ) . The Hilbert filter whereby the modulating signal
g (t ) can be converted to its orthogonal counterpart is easy to implement using digital filter design
tools. Moreover, the carrier amplitude can be adjusted based on the envelope of the modulating signal in practical applications. Thus, the total power consumption and the load of ultrasonic transducers could be greatly reduced 63.

g (t )
Hilbert
Transform

cos(ct )
sin(ct )

SSSBAM (t )

cos(ct )

Figure 4. Generation of SSBAM signal.

Subsequently, Kite et al. 64 predistorted the input modulating signal with a square-root operation to the envelope of DSBAM signal to compensate for the distortion brought about by the square
operation in the Berktays solution. This approach was later validated by Pompei 65 who devised a
practical parametric loudspeaker with low distortion. In Figure 5, it should be noted that the squareroot amplitude modulation (SRAM) produces a series of frequency components in the modulated
signal due to the square-root operation. However, experiments have revealed that SRAM technique
is effective to reduce the distortion to an acceptable level around 5% in practical use with certain
types of ultrasonic transducers 65.

SSRAM (t )

g (t )

+
1

()

x
cos(ct )

Figure 5. Generation of SRAM signal.

Another preprocessing technique involving quadrature modulation is modified amplitude

modulation (MAM), which was proposed by Tan. et al. 66 to offer a flexible bandwidth variety for a
wide range of ultrasonic transducers. From Figure 6, we can note that one branch modulation is the
same as DSBAM, while in the other branch the orthogonal carrier is modulated by a predistorted
term 1 m 2 g 2 (t ) to compensate for the harmonics generated in DSBAM. The main advantage of
MAM is that 1 m 2 g 2 (t ) can be substituted by its Taylor series of different orders to adapt to the
bandwidth of actual ultrasonic transducer. Performance analysis has been carried out in the reference 67, simulation results showed that MAM could achieve a similar THD level compared to
SSBAM and SRAM.
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21st International Congress on Sound and Vibration (ICSV21), Beijing, China, 13-17 July 2014

g (t )

()2

x
cos(c t )

m 2

sin(c t )

()

S MAM (t )

Figure 6. Generation of MAM signal.

The above-mentioned preprocessing techniques mainly focus on solving the harmonic distortion occurred in parametric loudspeaker systems, designers are therefore able to select the proper
preprocessing technique according to specific system requirements, such as analog or digital implementations, available computational resource, bandwidth limitation of ultrasonic emitter, etc. To
solve other limitation problems of parametric loudspeakers, researchers in recent year have concentrated their attention on enhancing the sound quality in low frequency range. In 2012, Shi et al. 68
introduced the missing fundamental concept widely used in virtual bass enhancement for conventional loudspeakers 69 to improve the bass quality for parametric loudspeakers. As shown in Figure
7, this psychoacoustical preprocessing technique splits the audio input with a low-pass and a highpass filter at a cut-off frequency of 500 Hz. The high-passed audio content is treated using the general preprocessing technique, while the low-passed bass is manipulated by function F ( x ) , which is
served as the harmonic generator for parametric loudspeakers. Subjective tests show that this preprocessing method could promote the bass quantity, and maintain a relatively good sound perception simultaneously.
Audio Input

Low-pass
Filter

Preprocessing
Function F ( x )

Parametric Loudspeaker

High-pass
Filter

Figure 7. Diagram of psychoacoustical preprocessing for bass enhancement in parametric loudspeakers.

Another method to improve the sound quality and equalize the frequency response of parametric loudspeakers was presented in the reference 70 by combining DSBAM and SSBAM techniques. In this work, the audio input signal is divided to three frequency ranges based on the characteristics of one commercial parametric loudspeaker. DSBAM is used to retain the sound energy in
the low frequency range below 2 kHz, while SSBAM deals with the 2-6 kHz components using
lower sideband and 6-10 kHz components using upper sideband, respectively. In addition, the
modulated signal is weighted to achieve a flat power spectrum of the audible sound in the output
terminal. Subjective tests demonstrate that this combined modulation method outperforms the single use of DSBAM or SSBAM in terms of sound loudness and quality, as well as the harmonic distortion.
4.2 Processing method based on Volterra series
Those Berktay-based preprocessing methods mentioned above cannot reduce the harmonic
distortions effectively when the levels of the primary waves are higher than a certain level due to
the limitation of the Berktays farfield solution 67. The nonlinear mechanism in parametric loudspeakers can also be modelled by certain types of filters with nonlinear structures. One versatile
approach for characterizing the nonlinear process is to utilize the Volterra series with a straightforward filter structure 71. In 1995, the nonlinear process in horns and ducts was investigated by Klip-

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21st International Congress on Sound and Vibration (ICSV21), Beijing, China, 13-17 July 2014
pel 72 using a finite Volterra series expanded to the nth-order kernel. Subsequently, a simplified
Volterra filter with one-dimension coefficient vector was proposed by Farina et al. 73 in 2001, and
has been successfully applied to nonlinearity modelling of an audio limiter 74, tube preamps and
musical instruments 75.
As the Volterra series is advantageous in nonlinear system identification, and the nonlinear
distortion of the parametric loudspeaker system is mainly attributed to the second harmonic element,
an adaptive Volterra filter with the 1st- and 2nd-order kernels in a cascaded structure has been
adopted by Ji et al. 76 to model the nonlinearity of parametric loudspeaker systems. The output of
the parametric loudspeaker system y (n ) can be given as
y (n ) y1 (n ) y2 (n )

N1 1

N 2 1 N 2 1

m1 0

m2 0 m1 0

h1 (m1 ) x(n m1 )

h (m , m ) x (n m ) x (n m )
2

(3)

H1 x (n ) H 2 x (n ) ,

where x ( n ) is the input modulating signal, h1 ( m1 ) and h2 (m1 , m2 ) are the 1st- and 2nd-order kernels,
respectively. H1[] and H2 [] represent the 1st-order and 2nd-order kernel operators accordingly.
For theoretical analysis, Ji et al resorted to the KZK mathematical model to calculate the secondary sound as the desired signal assuming that the ultrasonic emitter has a flat magnitude response and zero phase shift. The input signal, which is a uniformly distributed random white noise
signal, modulates a 40-kHz carrier using DSBAM with m 0.8 to form the primary waves for investigating the quadratic nonlinearity presented in the secondary sound. Figure 8 illustrates the
steady state coefficients of the 1st- and 2nd-order kernels after 3.2104 iterations using the NLMS
adaptive algorithm, respectively. It can be observed that the 1st-order kernel approximates the impulse response of a distortionless transmission for linear output, while the 2nd-order kernel indicates that the quadratic nonlinearity is nearly memoryless with a dominant coefficient locating at
h2 (0,0) .

(a)

(b)

Figure 8. Coefficients of the (a) 1st-order and (b) 2nd-order Volterra kernels.

Conventional Volterra filters have been widely and efficiently applied to nonlinearity modelling of a large class of nonlinear systems. However, due to the complexity of higher order kernels,
Volterra series have been seldom implemented in practical parametric loudspeaker systems. One
important challenge encountered in real-time system is the requirement of fast system modelling
and inverse filter design for predistorting the input signal. To address the issue of high computational complexity, a simplified Volterra filter with one-dimension kernel has been proposed and
studied in the reference 77. Generally, the one-dimension Volterra series for characterizing a discrete nonlinear system can be described as

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21st International Congress on Sound and Vibration (ICSV21), Beijing, China, 13-17 July 2014

y (n )

N1 1

h1 (m) x(n m)

m 0

N 2 1

h (m)x (n m)

m 0

(4)

N p 1

h p (m) x (n m)
p

m 0

where hp (m) is defined as the pth-order one-dimension Volterra filter kernel, and N p is the kernel
length. For parametric loudspeaker system modelling, Equation (4) is truncated up to the 2nd-order
kernel with p 2 .
To obtain the values of the 1st- and 2nd-order one-dimension kernel, an exponential swept-sine

ln( 2 )
signal x(n) A sin 1T / ln( 2 ) e N 1 1 is used as the input signal, where A is the amplitude,

1 and 2 are start frequency and stop frequency, respectively, T is the signal length, and N is
the sample number. The values of 1 and 2 should be carefully selected to meet the demand of
the dynamic behaviour of the nonlinear system within the sweeping range, and the value of T must
be long enough for an acceptable signal-to-noise ratio (SNR).
For a parametric loudspeaker system excited by the exponential swept-sine signal, the system
linear and quadratic impulse responses, namely h1 ( n ) and h2 ( n ) can be obtained by convolving
y (n ) with the inverse of x ( n ) , h1 ( n ) and h2 ( n ) are calculated using the following formula as

h1 (n) h1 (n)
(5)
,

h
(
n
)

2
h
(
n
)
/
A

2
2
The modelling performances of the conventional Volterra filter and the simplified onedimension Volterra filter have been evaluated by measurements. Seven input monotones from 0.5
kHz to 3.5 kHz with an increment of 0.5 kHz are passed to an actual parametric loudspeaker system
and the two Volterra models, respectively. The performance index is defined as the relative error
E m , which is given as
H i [ x ] Yi
100%, i 1, 2 ,
(6)
Yi
where i represents the harmonic order, Yi is the ith harmonic existing in the output of the actual
parametric loudspeaker, and H i is the output of the ith-order Volterra kernel. Figure 9 shows the
Ei

curves of relative error Ei using DSBAM for the conventional Volterra filter and the onedimension Volterra filter. It can be noted that the modelling errors for both the 1st- and 2nd-order
kernels of the conventional Volterra filter could be controlled around 5%, which indicates that the
Volterra model is effective to predict the output sound intensity under the same experimental conditions. For the one-dimension Volterra kernels, the linear output could exhibit the similar performance, while the relative errors of the quadratic output using the simplified 2nd-order kernel increase
to around 10% due to the approximation error.

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10

21st International Congress on Sound and Vibration (ICSV21), Beijing, China, 13-17 July 2014
35

Conventional Volterra filter 1st-order

Conventional Volterra filter 2nd-order
One-dimension Volterra filter 1st-order
One-dimension Volterra filter 2nd-order

Relative Error (100%)

30
25
20
15
10
5
0
0.5

1.5

2
2.5
Frequency (kHz)

3.5

Figure 9. Modelling errors of Volterra kernels using DSBAM.

5.

Implementation issues

In this section, the implementation issues about developing the parametric loudspeakers in
analog and digital circuits are summarized. In general, the parametric loudspeaker system consists
of three main parts, namely signal processing, amplifier, and ultrasonic emitter, as shown in Figure
10. Initially, parametric loudspeakers were based on analog circuits 2, 78, especially the parts of signal processing and amplifier shown in Figure 11. The signal processing is divided into two steps.
Firstly, the input signal is fed by the peak level detector, which can be designed with a voltage
comparator (LM111). The purpose of the peak detector is to avoid the overmodulation and get a
positive the envelop signal exported from adder that is simply achieved by using an operational
amplifier. Afterwards, the envelop signal is preprocessed by square-root circuit implemented by a
four-quadrant analog multiplier (AD734). This preprocessing is employed to reduce the distortion
produced during the nonlinear interaction in the air propagation, which have been explained in Section 3. Finally, the preprocessed signal is modulated by the carrier signal through the multiplier. The
carrier signal is produced by MAX038, which is a high-frequency, precision waveform generator.
Although the analog development of a parametric loudspeaker has the advantages of simple structure and low cost, it is lack of stability and flexibility. Furthermore, it is not suitable to implement
complicated functions and operations, such as Hilbert filter in the SSBAM scheme and dynamic
range control. These unfavourable factors have limited the development of high performance parametric loudspeaker.
Input Signal

Signal
Processing

Power
Amplifier

Ultrasonic
Transducer

Primary
wave

Figure 10. Block diagram of parametric loudspeaker.

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21st International Congress on Sound and Vibration (ICSV21), Beijing, China, 13-17 July 2014

+15V

LM111

Input

Envelope signal

OP27
-15V

Peak level detector

Multiplier

Carrier generator
Frequency

Z1

REF
DADJ

AD734

MAX038

Z2

IIN
FADJ

X1

Modulated signal

AD734
Y1

OUT

Figure 11. Analog implementation of parametric loudspeaker system.

Compared with analog circuit, digital circuit is more flexibility in complicated algorithms
without incurring extra cost, size, and power consumption. With the advantage of flexible configuration and high performance, Field Programmable Gate Array (FPGA) is a good choice to implement the parametric loudspeaker. The Contemporary FPGA has large resources of logic gates and
RAM blocks to implement complex digital computations. FPGA is able to perform parallel operations, as well as serial operations. The operation is chosen depending on the complexity of the algorithm and resource size of FPGA.
The generic block diagram for implementing digital preprocessing and modulation in the parametric loudspeaker is shown in Figure 12 79. The digital processing is divided into four steps: dynamic range control (DRC) processor, pre-distortion processing, automatic carrier level controller,
and amplitude modulator. The DRC processor consists of a compressor and a limiter. The compressor amplifies quiet sounds or reduces the volume of loud sounds by narrowing an audio signals
dynamic range. The limiter is employed to prevent the sound from clipping that will bring perceivable distortion of the difference frequency sound. The pre-distortion processing has been previously
discussed in Section 3, which reduces the amount of harmonic distortion and intermodulation distortion. In the third step, the sinusoidal oscillator generates a carrier signal, and automatic carrier
level control produces a dynamical gain. The genetic implementation ways of sinusoidal oscillator
include recursive algorithm, a look-up table and direct digital frequency synthesis (DDS). The dynamic carrier signal, which significantly reduces the power consumption, is obtained by multiplying
the carrier signal with dynamical gain. In the final step, the input signal is modulated by the dynamic carrier signal.

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21st International Congress on Sound and Vibration (ICSV21), Beijing, China, 13-17 July 2014
Digital Processing using FPGA

Sinusoidal
Oscillator

Input

Dynamic Range
Control Processor

Pre-distortion
Processing

Automatic Carrier
Level Controller

Amplitude
Modulator

Output

Figure 12. Digital processing of parametric loudspeaker.

Power
Supply

FPGA Board

+5V &+3.3V

Interface Board

Power
Regulator

Reset
LEDs
Crystal
Oscillator

FPGA

Switch
signal
Analog
Output

Clock
signal
Audio Codec
Data

Low-pass
Filter

Analog
Input

Figure 13. Interconnection between FPGA board and interface board.

Figure 13 shows the connecting diagram between the interface board and the FPGA chip
board. The FPGA chip performs the digital signal processing. The interface board consists of an
audio codec, voltage regulators and low-pass filters. The audio codec provides a two-channel 16-bit
96 kHz analog-to-digital converters (ADCs) and 192 kHz digital-to-analog converters (DACs) with
single-ended analog input/output. As the resonant frequency of the most ultrasonic emitters used for
parametric loudspeaker is between 30 kHz and 50 kHz, the bandwidth of 192 kHz DAC is wide
enough. In order to reduce the harmonic distortion of analog signals from DAC, low-pass filters are
essential. In addition, Voltage regulators produce 3.3 V and 5 V power supply.
In order to create parametric array effect in air, the modulated signal must be power amplified
before exporting to the ultrasound emitter. Commonly, the power amplifiers can be divided into
types of Class-B, Class-C, and Class-D. The last one is widely used in the parametric loudspeaker
due to its high efficiency and small size. Figure 14 gives a scheme of Class-D amplifier used in the
parametric loudspeakers, in which the pulse modulator can be implemented by the ways of analog
circuit and digital function. Compared to analog circuit, pulse train can be directly converted from
the digital signal by digital function without using the DAC. In the term of pulse modulation, there
are two common techniques: pulse density modulation (PDM) and pulse width modulation (PWM).
PWM generates variable duty pulses, the widths of which are proportional to the input amplitude;
PDM generates variable density pulses, the density of which corresponds to the input signals amplitude 80. PDM generates less harmonic distortion than PWM, but has lower efficiency. In the
Class-D output stage block, the PWM signals are amplified by power MOSFET transistors to drive
the ultrasound transducer. The level of amplified PWM signal reaches the supply voltage. Due to
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21st International Congress on Sound and Vibration (ICSV21), Beijing, China, 13-17 July 2014
the presence of dead time during the switching MOSFET transistor, the distortion and noise are
brought into the signal. In order to significantly alleviate the problem, an error correction technique
is required 81. Finally, an EMI filter, which should be designed elaborately to remove the high frequency products and maximize the power transfer, is needed before driving the transducers.

Input Signal

Demodulation
Filter

Pulse
Modulator

Output
Analog

Output Stage

Figure 14. Block diagram of Class-D amplifier for parametric loudspeaker.

The steerable parametric loudspeaker consists of several sub parametric loudspeakers, each
beam of which is delayed by specific time. The steering angle is determined by the value of delay.
Figure 15 shows the system structure of the steerable parametric loudspeaker. The digital processing algorithm is basically described in Figure 12, whereas the realization approaches of delay are
various. Olszewski et al. realized the system of steerable parametric loudspeaker with hybrid of
mechanical tilting and electronic time delays 82. The whole system includes four sub-arrays, which
are controlled by separate stepper motors. This method does not bring in grating lobes, but results in
the complexity and bulk of the system. Yang et al. achieved the constant beamwidth beamforming
of the steerable parametric loudspeaker using the Chebyshev weighting function and confirmed in
the simulation experiments 83. Gan et al. proposed an algorithm that the carrier and sideband frequencies were separately delayed 3. Owing to the main lobe of sideband frequencys directivity is
almost flat, the difference frequency directivitys steering angle and main lobe beamwidth were
inherited from the carrier frequencys directivity. By combining digital loudspeaker technique,
Takeoka et al. individually modulated and delayed every element signal of the steerable parametric
loudspeaker to emit multiple independent audio beams 84. Also, Wu et al. used fractional delay algorithm to steer the beam of parametric loudspeaker to any arbitrary angle 85. Since the spatial sampling theorems are not satisfied, the difference frequency beam presents grating lobes. Pompei et al.
presented a method that utilized oversized array transducers carefully arranged to eliminate grating
lobes 86. Shi et al. proposed a Gaussian source array in place of the ultrasonic transducer array to
prove the feasibility of devising a digital beamsteerer for the parametric loudspeaker 87. According
to Gaussian source array, the directivity characteristics of linear ultrasonic transducer array were
analyzed. Using the delay and Chebyshev weighting, the grating lobe of difference frequency was
suppressed. Lee et al. proposed a way to versatile beamforming, using the complex weights in place
of time delays 88. In the term of beamsteering simulation, Shi et al proposed three modified product
directivity models to predict the locations and amplitudes of the mainlobe and grating lobes 87.
Compared with the original product directivity model, the models proposed by Shi promoted the
selected sidelobe predictions by about 10 dB.

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21st International Congress on Sound and Vibration (ICSV21), Beijing, China, 13-17 July 2014

ADC

Audio

Delay1

DAC1

Drive1

Delay2

DAC2

Drive2

Delayn-1

DACn-1

Driven-1

Delayn

DACn

Driven

Digital
Processing

Figure 15. Block diagram of steerable parametric loudspeaker.

6.

Applications

Parametric loudspeaker is capable of creating directional sound beam using an ultrasonic

emitter, which has been used in personal entertainment system, communication, and messaging
system 89. In this section, some recent applications are listed which show good potential in exploiting unique features of parametric loudspeaker.
Traditionally, omni-directional loudspeakers are used as control sources to suppress the noise in
the targeted area in the active noise control (ANC) system. However, the sound pressure level (SPL)
of the noise at locations away from the target area may actually increase. Brooks et al. 90 investigated the feasibility of using a parametric loudspeaker as a control source in the ANC system. Experimental results showed that the ANC system using parametric loudspeaker could reduce the
SPLs at the control points without increasing noise level in other areas. Kidner et al. 91 proposed a
method of combining the parametric loudspeaker and virtual sensing techniques to create localized
zones of quiet. Tanaka and Tanaka 92-93 presented a novel idea of using a steerable parametric loudspeaker based upon phased array theory with an optimal control law to track a moving target point
without rotating the parametric loudspeaker mechanically. Furthermore, the same research group 94
proposed a novel method to use the reflected sound wave, which was produced by the parametric
focusing loudspeaker on the target source, to collocate both the primary source and the control
source. Therefore, global noise control can be achieved because the distance between the two
sources became theoretically nullified. In another work, Komatsuzaki and Iwata 95 compared the
interfered sound field of using the parametric loudspeaker with omni-directional loudspeaker as
control sources to achieve local noise control. Three different constructions of the primary source
and the control source were considered: opposed placement, orthogonal placement, and coaxial
placement. Results showed that the parametric loudspeaker can be used as a control source to mitigate sound pressure level locally without increasing noise level in other areas.
In the area of personal communication, Nakashima et al. 96-97 mounted two parametric loudspeakers on a prototype mobile phone to deliver private sound field (or personal sound) to the users.
Each parametric loudspeaker consists of 16 piezoelectric transducers. The SPL of the audible sound
on the central axis was measured to be more than 70 dB at a distance of about 50 cm away from the
phone. Because of the directivity of parametric loudspeaker, the sound pressure difference between
the two ears is approximately 15 dB, which is helpful in binaural sound reproduction.
Due to the high directivity of sound waves at low frequency ranges, parametric loudspeaker has
also been found to be suitable in detecting concealed objects, such as land mines or weapons. Haupt
and Rolt 98 used the parametric loudspeaker to excite the buried mines from a safe distance away.
The vibration signatures of the mines were measured by the laser vibrometer to locate the position
of the mines. In another work, Achanta et al. 99 presented a novel method to detect concealed weapons by parametric loudspeakers. The metallic and non-metallic materials under clothing, as well as
the abnormality in handheld devices, can be scanned and detected by the directional sound at low
frequencies generated by the parametric loudspeaker system.

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Sayin et al. proposed a method of utilizing parametric loudspeaker to build an omni-directional
source of sound, which consists of a sphere with hundreds of ultrasonic transducers on its surface.
Due to the parametric loudspeaker phenomenon, the source showed highly omni-directional for the
mid-high frequency range. Moreover, the spatial correlation of the generated diffuse sound by the
omni-directional parametric source diminished to zero more rapidly than using a dodecahedron
source. His group also proposed a parametric loudspeaker with horns, which had obvious effects on
the efficiency and directivity of the emitted sound.
Parametric loudspeaker can also be deployed to measure the acoustic performance of materials. Bernards group used a parametric loudspeaker and a microphone to measure the absorption
coefficient of materials at normal incidence in air under the excitation of impulse signals 100-101.
Kuang et al. introduced the transfer function method into the process 102, which allowed measurements to be taken in situ. Moreover, this method can be combined with the phase-cancellation
method proposed by Kamakura60 to effectively remove spurious noise for accurate measurements103.

7.

Conclusions

This review paper outlines some of the important theoretical and experimental developments
of parametric loudspeaker. An overview of numerical modelling for nonlinear acoustics, differencefrequency sound measurement, signal processing techniques, implementation issues and its applications for parametric loudspeakers is provided. There still exist some technical challenges to be
overcome. For instance, although many preprocessing methods have been proposed to improve the
quality of the demodulated signal, new method is still preferred, especially in the low frequency
ranges. There is still a long way to go to develop a parametric loudspeaker with lower power consumption and a better quality.

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