IEEE TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 19, NO.
6, AUGUST 2011
Conversion of Multichannel Sound Signal
Maintaining Physical Properties of Sound
in Reproduced Sound Field
Akio Ando, Member, IEEE
Abstract—In this paper, we describe a new method for con-
verting the signal of the original multichannel sound system into
that of an alternative system with a different number of channels
while maintaining the physical properties of sound at the listening
point in the reproduced sound field. Such a conversion problem
can be described by the underdetermined linear equation. To
obtain an analytical solution to the equation, the method parti-
tions the sound field of the alternative system on the basis of the
positions of three loudspeakers and solves the “local solution” in
each subfield. As a result, the alternative system localizes each
channel signal of the original sound system at the corresponding
loudspeaker position as a phantom source. The composition of the
local solutions introduces the “global solution,” that is, the ana-
lytical solution to the conversion problem. 22-channel signals of
a 22.2 multichannel sound system without the two low-frequency
effect channels were converted into 10-, 8-, and 6-channel signals
by the method. Subjective evaluations showed that the proposed
method could reproduce the spatial impression of the original
22-channel sound with eight loudspeakers.
Index Terms—Acoustic signal processing, sound field repro-
duction, sound localization, spatial and multichannel audio,
three-dimensional sound.
I. INTRODUCTION
B ROADCASTING and packaged media have popularized
5.1 multichannel sound as a home sound system. Research
on multichannel audio has now shifted to advanced systems with
more channels to provide enhanced spatial impressions. For ex-
ample, a 22.2 multichannel sound system has been developed
for ultrahigh-definition television with 4320 scanning lines [1].
Such advanced sound systems require their own loudspeaker ar-
rangement to bring out the best performance. Although loud-
speakers can be arranged optimally in a theater, they are diffi-
cult to set up in a typical home environment.
“Down-mixing” is a widely known method of reducing the
number of channels in multichannel audio. Down-mixing from
5.1 to two-channel stereo or monophonic has already been stan-
dardized in an ITU-R Recommendation [2] and is applied to
Manuscript received June 29, 2010; revised October 12, 2010 and November
01, 2010; accepted November 02, 2010. Date of publication November 15, 2010;
date of current version May 25, 2011. The associate editor coordinating the re-
view of this manuscript and approving it for publication was Prof. Lauri Savioja.
The author is with NHK Science and Technology Research Laboratories,
Tokyo 157-8510, Japan (e-mail: ando.a-io@nhk.or.jp).
Color versions of one or more of the figures in this paper are available online
at http://ieeexplore.ieee.org.
Digital Object Identifier 10.1109/TASL.2010.2092429
1558-7916/$26.00 © 2010 IEEE
1467
some television receivers. Although such a down-mixing algo-
rithm was reported as having a certain effectiveness [3], it does
not work for an arbitrary loudspeaker arrangement.1 To enable
down-mixing between a number of systems, a technology for
converting or recreating sound fields is necessary.
A sound field reconstruction method was proposed in the
1960s [4]. It tried to recreate the primary sound field of a
large room, such as a concert hall, in a small room (the sec-
ondary sound field) based on Huygens’ principle, which can
be regarded as an approximation of the Kirchhoff–Helmholtz
integral theorem [5]. There are other methods using the Kirch-
hoff–Helmholtz integral theorem directly [6], [7]. In those
methods, it is necessary to enclose the listening area by a
boundary surface, on which sound pressure and its gradient
are controlled. Another method of sound field reconstruction
is wave field synthesis (WFS) [8], [9]. In WFS, the boundary
surface enveloping the listening area degenerates to a plane,
and then, it uses a Rayleigh I or II integral instead of the
Kirchhoff–Helmholtz integral [10]. However, those sound
field reconstruction methods require many loudspeakers or a
loudspeaker array in order to satisfy their physical theory.
On the other hand, there are some methods that can be
used to reproduce the original sound pressure at a listening
point, instead of the listening area that has a certain volume.
Among them, ambisonics is a widely accepted sound pickup
and reproduction method [11], and higher order ambisonics
[12], [13] can expand the listening area if a certain number of
secondary sources (loudspeakers) are available and uniformly
arranged [14]. Ambisonics represents the observed sound
pressure by spherical harmonic expansion with a finite number
of components up to a given order and tries to reproduce the
sound that has the same expansion coefficients through several
loudspeakers surrounding the listening point or area. It offers
a complete hierarchical approach to sound pickup, transmis-
sion and reproduction, which corresponds to mono, stereo,
horizontal surround, or three-dimensional sound reproduction.
However, it has the problem of the unavailability of higher order
sound pickup devices, and even with the specially designed
microphone array [15], the expansion in actual ambisonic
encoding is still limited up to several orders at present.
All sounds reproduced can be considered binaural because
they are heard by both ears of a listener. Thus, the binaural
recording with the so-called “dummy head” could bring the spa-
tial sound impression of the original sound field to the listener.
1The down-mixing method standardized in [2] uses only three down-mixing
coefficients, i.e., 1.0, 0.7071, and 0.0.
1468 IEEE TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 19, NO. 6, AUGUST 2011
Fig. 1. Block diagram of proposed method.
TABLE I
REQUIREMENTS FOR SOUND REPRODUCTION
Moreover, there are some methods to reproduce binaural signals
on loudspeakers [16], [17]. However, a mismatch of the head-re-
lated transfer function (HRTF) [18] between the dummy head
and the listener’s real head degrades the spatial impression of
the reproduced sound.
The conventional methods mentioned above aim at recreating
the primary sound field, such as that of a live performance in a
concert hall, in a different room. Such a problem is difficult and
very challenging, as it had been thought as “impossible” [4]. On
the other hand, the problem of recreating the original “repro-
duced sound field,” such as the sound field in the mixing studio
equipped with a multichannel sound system, into another room
seems to be easier to be solved than the conventional problem.
However, only a few attempts have so far been made regarding
this problem, which is the purpose of this study. In this paper, the
sound field reproduced by the original sound system is referred
to as the “original” sound field and the sound field recreated
by an alternative system with a different number of channels
is referred to as the “reproduction” sound field. The problem
of recreating the primary sound field, such as that in a concert
hall, is the task of a recording engineer and is not dealt with in
this paper. Our problem of recreating the original sound field in
another room corresponds to that of finding a good conversion
from the sound signal of the original sound system to that of the
reproduction sound system.
In such conversion, we aim for the faithful reproductions of
1) timbre, 2) sound localization, and 3) the sound envelopment
of the original sound [19]. Such requirements are shown in
Table I. Regarding requirement 1), the conversion discussed
in this paper should not be a function of frequency, because
a frequency-dependent conversion might change the timbre.
Among the conventional methods, ambisonics [11]–[13] seems
to meet requirements 1) and 3). However, ambisonics cannot
prevent the generation of a negative conversion coefficient
that brings out an opposite-phase signal that would degrade
the localization of the reproduced sound [20]. Consequently,
we developed a new conversion method satisfying all three
requirements.
In this paper, a new conversion method that is applicable to
automatic down-mixing and up-mixing is presented. The basic
concept of this method is to solve the problem on the conversion
matrix that converts the input signals for loudspeakers in the
original sound field into those in the reproduction sound field in
such a way that the physical properties of sound coincide at the
receiving point (listening point). This problem can be solved an-
alytically and its solution is frequency-independent, indicating
that the conversion does not change the timbre.
II. FORMULATION OF SOUND FIELD CONVERSION
The proposed method converts the original signal into
the converted signal through matrix operation:
(1)
where
.. ..
. .
The matrix should be solved such that the physical proper-
ties at the receiving point in the original and reproduction sound
fields are the same. Fig. 1 shows the block diagram of the pro-
posed method, where denotes the sound propagation from
loudspeakers to the receiving point in the original field and
denotes that in the reproduction field. If a physical property of
sound is a linear function of the audio signal, the equation that
should satisfy becomes
(2)
could be formally written as
Equation (2) can be solved with numerical algorithms. However,
the presented approach tries to solve it analytically.
Because a physical property of sound is a function of location
in a three-dimensional space, its dimension is at most three. We
select the three-dimensional property of sound in this paper. As
ANDO: CONVERSION OF MULTICHANNEL SOUND SIGNAL MAINTAINING PHYSICAL PROPERTIES OF SOUND IN REPRODUCED SOUND FIELD
a result, becomes a matrix. In such a case, (2) be-
comes the underdetermined linear equation if , which is
the usual situation in a three-dimensional sound field reproduc-
tion. To obtain an analytical solution, we divided the matrix
into 3 3 partial matrices and solved each partial problem. The
physical meaning of this division is to partition the reproduction
field on the basis of the positions of three loudspeakers. Using
the linearity of the physical property of sound again, the conver-
sion matrix is further divided into the problems of conversion
from one channel to three channels. The solution to this con-
version problem is called the “local solution” in this paper. In
contrast, the analytical solution , which is composed of local
solutions, is called the “global solution.” The physical meaning
of the local solution is to create a phantom source using three
loudspeakers in the reproduction field. Since the restriction on
and is only , the method can be applicable to both
down-mixing and up-mixing.
The following assumptions are made: 1) each loudspeaker
can be modeled as a point source; 2) the sound pressure at a unit
distance from a loudspeaker is proportional to the input to the
loudspeaker (the proportionality coefficient is denoted as );
3) only the outgoing wave from the loudspeaker is considered;
and 4) the reflected sound can be neglected in the original and
converted fields. Furthermore, 5) is assumed, where
is the wave number and is the minimum distance between
the loudspeakers and the receiving point. Assumption 5) is valid
except for the low-frequency sound that does not contribute to
the perception of sound localization.
It is well known that sound can be described by the following
two physical properties: sound pressure and particle velocity
[21]. Sound pressure is a scalar variable of time and location. On
the other hand, particle velocity is generally a three-dimensional
variable. Note that both physical properties are linear functions
of the source signal. On the basis of the assumptions mentioned
above, if a loudspeaker whose input signal is is located at
, the Fourier transforms of sound pressure and
particle velocity at the receiving point can be
written as
(3)
and
(4)
respectively, where is the imaginary unit, superscript de-
notes the transposition of the vector or matrix, and is the pro-
portionality coefficient stated in the preceding paragraph. We
used assumption (5) to derive the last row of (4).
Suppose there are loudspeakers in the original sound field,
whose locations are , , and
1469
Fig. 2. Generation of phantom source at  (position of loudspeaker in original
field) with three loudspeakers located at  ,  , and  in reproduction field.
loudspeakers in the reproduction sound field, whose locations
are , . Let the input signals
of the loudspeakers in the original space be , ,
and those in the reproduction space be , ; the
Fourier transforms of sound pressure and particle velocity at the
receiving point in the original sound field are represented as
(5)
(6)
and those in the reproduction sound field as
(7)
(8)
To obtain an analytical solution in which the physical prop-
erties of sound coincide at the receiving point, the proposed
method finds a local solution creating a phantom source of an
original loudspeaker at the corresponding position in the repro-
duction field. The phantom source is generated by three loud-
speakers, whose directions make a spherical triangle including
the source direction. Fig. 2 shows this step of the method. The
method then creates a global solution by summing up such
local solutions for all the loudspeakers of the original field based
on the linearity of the physical property of sound.
III. ANALYTICAL SOLUTION
A. Selection of Three Loudspeakers in Reproduction Field to
Generate Phantom Source in Original Field
We shall use polar coordinates originating at the receiving
point and represent a position as , where is the distance
from the origin, is the azimuthal angle, and is the elevation
angle. When only the direction is considered, the position is
represented as in this paper.
To create a phantom source as shown in Fig. 2, three loud-
speakers in the reproduction field should be selected for each
loudspeaker position in the original field. Only the loudspeaker
direction is considered here. We assume that a loudspeaker in
the original field is located at . Three loudspeaker positions
1470 IEEE TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 19, NO. 6, AUGUST 2011

in the reproduction field should be selected, satisfying the fol- A virtual source of the original loudspeaker is localized at
lowing conditions: 1) a spherical triangle formed by the three in the reproduction space if
loudspeaker directions includes the direction of , and 2) the
spherical area of the triangle is the minimum among those sat- (13)
isfying 1). holds. A local solution can be then obtained as
Using the notation in Fig. 2, we define three vectors:

(14)

(9) where
where denotes the vector product. If and only if the position
is inside the spherical triangle determined by , , and , the
three vectors , , and have the same direction. Therefore,
we can determine that the direction of is inside the triangle if

(10)

holds, where denotes the inner product of two vectors. We used

this method to verify condition 1).

B. Local Solution That Generates Phantom Source

Assuming that a loudspeaker is located at in the orig-
inal space and its input signal is , the Fourier transform of
the particle velocity observed at the receiving point is

(11)

where is the density of air, is the speed of sound, and

Solution (14) guarantees the coincidence of the particle ve-
locities at the receiving pont. Now, let us introduce a condition
for the coincidence of sound pressures. In the current case, the
Fourier transform of the sound pressure at the receiving point
Here, is the distance between the source and the receiving due to a loudspeaker at is
point, is the azimuthal angle, and is the elevation angle. On
the other hand, if there are three loudspeakers in the reproduc- (15)
tion space, each of whose input signal and location are
and , respectively , the Fourier trans- and that due to the three loudspeakers at ,
form of the observed particle velocity is is

(12)
(16)
where we have the equation shown at the bottom of the page,
and From (15) and (16), the coincidence of sound pressures is
obtained when

(17)
ANDO: CONVERSION OF MULTICHANNEL SOUND SIGNAL MAINTAINING PHYSICAL PROPERTIES OF SOUND IN REPRODUCED SOUND FIELD
is true. The solution does not satisfy condition (17) because
substituting (14) into (17) yields an incorrect equation:
On the other hand, a new vector
(18)
satisfies condition (17), where
(19)
Although solution (18) does not satisfy condition (13), it guar-
antees the coincidence of the directions of particle velocities.
Therefore, solution (18) gives the same sound pressure and
sound direction in both fields.
The obtained solution depends not on the input signal but on
the positions of the loudspeakers and receiving point in both
fields. One of the obtained signals is
in the frequency domain. The time-domain representation is
(20)
From (20), it is clear that the conversion is frequency-indepen-
dent and satisfies requirement 1) stated in Section I.
The problem to create a phantom source is known as “pan-
ning” [22]–[26]. Among the panning methods, vector-based
amplitude panning (VBAP) [25], [26] is widely regarded as a
promising method of three-dimensional panning. VBAP could
be considered a generalization of the “tangent law” [22] of a
three-dimensional loudspeaker setup. If all distances between
loudspeakers and the receiving point shown in Fig. 2 are the
same, local solution (14) yields the same result as VBAP. Thus,
solution (14) provides physical underpinnings for the use of
VBAP [27].
If we select only particle velocity as the physical property of
sound, it is necessary to “normalize” solution (14) to control the
loudness of the reproduced sound. Actually such normalization
was introduced in VBAP [25]. In the panning problem, there
is little restriction on such normalization. However, in the mul-
tichannel sound conversion problem considered in this paper,
the normalization of each local solution loses the loudness bal-
ance among phantom sources and fails to compose the global
solution. On the other hand, solution (18) normalizes (14) in a
natural way that coincides with the sound pressure. Thus, we
adopted solution (18) as a local solution.
C. Global Solution
A global solution ( matrix) can be obtained by
putting each element of local solutions into its appropriate por-
tion. Let the loudspeaker of the th channel in the original space
1471
Fig. 3. 22.2 multichannel loudspeaker arrangement without LFE channels.
be included in the area bounded by the three loudspeakers of the
th, th, and th channels in the reproduction field.
The elements of the local solution in this case are denoted as
Using this notation, we define as
0
where . Accordingly, the matrix
becomes the global solution because the sound
pressure and particle velocity are linear functions of the input
signal.
Instead of sound pressure and particle velocity, we could use
sound intensity [28] as the physical property of sound and obtain
an alternative solution of (18). Although doing so is valid for
the local solution, the global solution cannot be obtained by the
simple addition of local solutions, because sound intensity is a
quadratic variable of the original signal. This is the reason we
did not select sound intensity as the physical properties of sound.
IV. SUBJECTIVE EVALUATION
Two subjective experiments were carried out in order to eval-
uate the proposed method. In the first experiment, a 22.2 sound
signal without the two low-frequency effect (LFE) channels was
converted into a 10-channel signal. The converted sound was
compared with the original sound by subjective evaluation. In
the second experiment, a 22-channel signal was converted into
8- and 6-channel signals, and the converted sound was compared
with the original sound. Fig. 3 shows the 22.2 multichannel
loudspeaker arrangement used in the experiments. All the ex-
periments were carried out in a soundproof room where the re-
verberation time at 500 Hz was 0.18 s. The distance between the
listening point and each loudspeaker was 2 m.
Eight sound stimuli were obtained from 22.2 multichannel
programs exhibited at World Expo 2005 held in Aichi, Japan,
and NAB 2006 and 2007 shows held in Las Vegas, NV. Each
stimulus was from 10 to 12 s long. The stimuli included musical
and sport sounds, birds singing, and the sound of a light breeze.
1472 IEEE TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 19, NO. 6, AUGUST 2011

TABLE III
RESULTS OF AUTOMATIC SELECTION OF THREE LOUDSPEAKER POSITIONS IN
REPRODUCTION FIELD WITH RESPECT TO EACH LOUDSPEAKER POSITION IN
ORIGINAL FIELD AND RESULTANT WEIGHTING COEFFICIENTS
Fig. 4. Double-blind triple-stimulus with hidden reference method.

TABLE II
SCALES USED FOR SUBJECTIVE EVALUATION

Fig. 5. Ten-channel loudspeaker arrangements in reproduction space.

A. Subjective Evaluation Method

Subjective evaluations used the “double-blind triple-stimulus
with hidden reference” method [29], which can be used for the
subjective assessment of small impairments in a multichannel
sound system. Fig. 4 shows the method. In Fig. 4, stimulus
“R” indicates the reference sound and stimuli “A” and “B” the
sounds for evaluation. The subject was asked to assess the im-
pairment on A and B compared with R, according to a contin-
uous five-grade impairment scale shown in Table II. Either stim-
ulus A or B was the same as R. The stimulus that is the same
as the reference is referred to as the “hidden reference,” and
the other stimulus the “object.” In the experiment, the reference
was the original 22-channel sound and the object the converted
sound. The impairment was assessed from the following two
viewpoints: sound localization and sound envelopment. Timbre
was not assessed in the experiments because in principle it could
be kept the same by the proposed method [please see (20)]. After
the experiment, the “difference grade” was calculated for each
object by subtracting the grade given to the hidden reference
from that to the object, and therefore, it should be a nonpositive
value.

B. Experiment-1: Conversion From 22-Channel Sound Signal

to 10-Channel Sound Signal
The three loudspeaker arrangements shown in Fig. 5 were
used to convert the 22-channel signal into a 10-channel signal.
Layout 1 had 4 loudspeakers in the top layer, 5 loudspeakers
in the middle layer, and 1 loudspeaker in the bottom layer.
Layout 2 had 3, 6, and 1 loudspeakers in the top, middle, and
bottom layers, and layout 3 had 3, 5, and 2 loudspeakers in
the top, middle, and bottom layers, respectively. The result
of the automatic selection of three loudspeakers (described
in Section III-A) and the resultant weighting coefficients (de-
scribed in Section III-B) for layout 2 are shown in Table III as
an example.
In the experiment, subjects were 38 people in their twenties,
thirties, and forties, who were experienced in playing musical
instruments. They evaluated each conversion twice. Eight sound
stimuli were used in the experiment.
The subjective evaluation results are shown in Fig. 6. P1, P2,
and P3 in Fig. 6 show the proposed method results for layouts 1,
2, and 3, respectively. D2 and D3 show the results of the conven-
tional down-mixing algorithm where each original signal was
converted with the coefficients of 1.0, 0.5, and 0.0 satisfying the
coincidence of sound pressures at the receiving point. Layout
1 had every other loudspeaker of the original 22-channel sound
system, and hence, the conversion matrix of the conventional
ANDO: CONVERSION OF MULTICHANNEL SOUND SIGNAL MAINTAINING PHYSICAL PROPERTIES OF SOUND IN REPRODUCED SOUND FIELD 1473

Fig. 6. Results of experiment-1. (a) Sound localization. (b) Sound envelop-

ment.
Mean scores and 95% confidence limits
P1: Proposed, layout 1
D2: Conventional, layout 2 P2: Proposed, layout 2
D3: Conventional, layout 3 P3: Proposed, layout 3

down-mixing algorithm was almost the same as that of the pro-

posed method for this layout. For example, both methods evenly
distributed the sound signal of a channel located in the (90, 0) di- Fig. 7. 8- and 6-channel loudspeaker arrangements in reproduction
rection in the original space to two channels located in the (120, space. (a) 8-channel loudspeaker arrangements; (b) 6-channel loudspeaker
0) and (60, 0) directions in the reproduction space. Because of arrangements.
this, the conventional down-mixing algorithm was not evaluated
for layout 1. The proposed method obtained a difference grade
of more than 1.0 for both spatial impressions. Thus, even with
10 channels, the proposed method kept the spatial impressions
of the original 22-channel sound and gave a better result than the
conventional down-mixing algorithm. The difference between
these methods was significant at a level of 0.05.

C. Experiment-2: Conversion From 22-Channel Sound Signal

to 8- and 6-Channel Sound Signals
In the first experiment, the proposed method obtained a differ-
ence grade of more than 1.0 in converting 22-channel sound
into 10-channel sound. The question arose as to how many chan-
nels in the reproduction space can keep a difference grade of
more than 1.0 for both spatial impressions. Hence, we con-
ducted the second experiment in which the original 22-channel
sound signal was converted into 8- and 6-channel sound signals.
We set up three loudspeaker arrangements for both conversions,
as shown in Fig. 7. The subjects were 32 people in their twen-
ties, thirties, and forties, who were experienced in playing mu-
sical instruments. They evaluated each conversion twice. Since
the number of loudspeaker arrangements in this experiment was
twice that in the first experiment, we considered four represen-
tative stimuli and used them in this experiment. The other ex- Fig. 8. Results of experiment-2. (a) Sound localization; (b) sound envelopment.
perimental conditions in this experiment were the same as those Mean scores and 95% confidence limits
in the first experiment. D1: Conventional, layout 1 P1: Proposed, layout 1
D2: Conventional, layout 2 P2: Proposed, layout 2
The results are shown in Fig. 8. Regarding layout 1 for the D3: Conventional, layout 3 P3: Proposed, layout 3
8- and 6-channel loudspeaker arrangements, the difference be-
tween the conventional down-mixing method and the proposed
area). On the other hand, the converted 6-channel sound did not
method is not significant at a level of 0.05. The reason for this
maintain a difference grade of more than 1.0.
is likely that both methods distributed the sounds of the front
channels of the original 22-channel sound in almost the same
V. DISCUSSION
manner. For the other layouts, the proposed method yielded
better results than the conventional down-mixing method did The proposed method guarantees the coincidences of pres-
on both spatial impressions, and the difference was significant sures and directions of sound only at the receiving point. It can
at a level of 0.05. The converted 8-channel sound in layouts 1 be extended into the method of minimizing the square errors
and 2 gave spatial impressions with a difference grade of more of those sound properties over a receiving area (an infinite set
than 1.0 (layout 3 had only a few loudspeakers in the frontal of receiving points) [30]. However, in such a case, the obtained
1474 IEEE TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 19, NO. 6, AUGUST 2011
solution will inevitably be a frequency-dependent function and
would not satisfy requirement 1) mentioned in Section I. Al-
though the proposed method ensures the coincidences of the
physical properties only at a receiving point, the subjective eval-
uation results showed that it is effective in a nearby area because
both ears would not be on the receiving point at the same time.
We shall study the size of the effective listening area reproduced
by the proposed method.
The down-mixing method proposed in [2] has coefficients
based on the conservation of sound power. On the other hand,
the conventional down-mixing method used in this study has co-
efficients based on the conservation of sound pressure, so as to
match the proposed method. The local solution of the proposed
method can be naturally made to conserve power by modifying
(15). However, in such a case, the global solution cannot be ob-
tained by adding the local solutions for the same reason as in
the coincidence of sound intensities.
Notwithstanding the above remarks, the down-mixing
method described in this paper performed very well. There
would be no reason for these results other than it guarantees
the coincidence of sound pressures. The principal difference
between the conventional and proposed methods is the repro-
ducibility of the sound direction. A subjective evaluation of
sound materials with a clear sound direction would increase the
difference between the two methods.
In this paper, we used subjective evaluation to evaluate the
spatial impression of sound. On the other hand, there are objec-
tive evaluation methods with interaural time difference (ITD),
interaural level difference (ILD), and interaural cross correla-
tion (IACC) [19]. Such methods could extract the spatial im-
pressions adequately with artificial sound sources, such as white
noise. However, it is difficult for such methods to evaluate the
impressions of a natural sound source that is a good test signal
for overall evaluation. Thus, we did not adopt the objective eval-
uation in this study. Both subjective and objective evaluation
methods with various test signals will be further studied in eval-
uation experiments.
In this paper, the proposed method was evaluated as
a down-mixing method. Obviously, it can be used as an
up-mixing method. We shall study this issue in due course.
VI. CONCLUSION
In this paper, we proposed a new method for converting
multichannel sound signals while maintaining the pressure
and direction of sound at the receiving point in the reproduced
sound field. We found that the conventional down-mixing
method would be effective if the sound pressure were con-
served. Even in such cases, the proposed method performed
better than the conventional method because it can reproduce
the sound direction. Subjective evaluations revealed that the
8-channel sound converted by the proposed method gave almost
the same spatial impressions as the original 22-channel sound
at a receiving point.
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ANDO: CONVERSION OF MULTICHANNEL SOUND SIGNAL MAINTAINING PHYSICAL PROPERTIES OF SOUND IN REPRODUCED SOUND FIELD
Akio Ando (M’80) received the B.S. and M.S.
degrees from Kyushu Institute of Design, Fukuoka,
Japan, in 1978 and 1980, respectively, and the
Dr.Eng. degree from Toyohashi University of Tech-
nology, Toyohashi, Japan, in 2001.
In 1980, he joined the Japan Broadcasting Corpo-
ration (NHK). He has been with the NHK Science
and Technology Research Laboratories, Tokyo,
Japan, since August 1983. He was in charge of
developing simultaneous subtitling systems for live
broadcast TV programs using speech recognition,
with which NHK started simultaneous subtitled broadcasting for daily news
1475
programs in March 2000 and sports and variety programs in December 2001.
Since 2002, he has been engaged in research on audio and acoustics including
acoustic signal processing, electroacoustical transducers, cognitive science
of acoustics, and spatial sound reproduction. From 2004 to 2006, he was the
Director of the Acoustics and Audio Signal Processing Division, and currently
he is a Senior Research Engineer of the Advanced Television Systems Research
Division, NHK Science and Technology Research Laboratories. Since 2010,
he has been a Guest Professor at the Tokyo Institute of Technology, Japan. His
research interests include pattern recognition, signal processing, theoretical
acoustics, and multichannel audio coding.