INTRODUCTION

Sound quality and noise annoyance are sometimes oversimplistically described by sound
pressure level or more complex psychoacoustic metrics that remain numbers only meaningful to
specialists (Powell and Fields, 1995). The possibility to conduct sound quality and noise
annoyance studies in laboratory conditions represents an interesting avenue for vehicles
manufacturers and researchers. Early work dedicated to sound reproduction or synthesis of
vehicles for sound quality or annoyance studies seems to be the NASA ANOSS project (McCurdy
and Grandle, 1987). Recently, Janssens et al. (2008) and Berckmans et al. (2008) investigated
analysis and synthesis algorithms for aircraft interior and exterior sounds. In these studies, the
spatial character of sound reproduction is not a primary aim. However, it is known that spatial
distribution of sound sources influences auditory masking and, therefore, the overall sound
quality. In our case, we focus on spectral and spatial quality of the sound. The aim is the
reproduction of an aircraft cabin sound field, namely, a target sound field, inside a full-scale
mock-up of the cabin.
In preliminary experiments (Gauthier et al., 2012), both the target and reproduced sound
fields were generated and measured in laboratory conditions: the target sound field was created
using four loudspeakers outside the mock-up to simulate an external excitation. Reproduction
was achieved using multichannel equalization based on pseudo-inversion and least-mean-square
methods with Tikhonov regularization to drive trim-panel actuators. In this paper, we
investigate similar types of algorithms but driven by real in-flight recordings. Effects of the
regularization parameter and crossover frequency on the objective performance are reported.
AIRCRAFT SOUNDS AND MULTICHANNEL EQUALIZATION
Except for transitional stages such as taxi, take-off, landing, engine starts, flaps operations,
most of the aircraft sounds as heard in passenger cabins are stationary, at least from frame and
perceptive viewpoint (Verron et al., 2011; Langlois et al., 2011). For fixed flight conditions and
aircraft configurations, the resulting sound environment is mostly made of emerging stationary
tones and filtered noise with stationary spectral envelop (Ploner-Bernard et al., 2005; Verron
et al., 2011). Tones result from rotating machinery while broadband noises are created by the
turbulent boundary layer on the fuselage, engine jet noise and the air climate system (Mixson
and Wilby, 1995; Wilby, 1996). In this paper, we investigate sound field reproduction of these
stationary sounds measured in-flight. To illustrate these signal assumptions, a typical aircraft
sound as recorded at a single position in a Bombardier CRJ900 is shown in Fig. 1. Note that
absolute physical units are deliberately not presented in this paper.
As shown in Fig. 1(a), the sound pressure level is stationary. Figs. 1(b) and (c) show the
overall loudness (Zwicker and Fastl, 1999) as function of time and the signal spectrogram. On
the basis of the signal spectrogram, the aforementioned assumptions are verified. Although the
sound field reproduction methods and multichannel signal processing presented in this paper do
not rely on signal stationarity hypothesis, the fact that the investigated sound fields are
stationary circumvents known issues encountered in single channel or multichannel
equalization algorithms based on system inversion, i.e. audible artifacts such as pre- and
post-echoes (Gauthier et al., 2012) or ringing. However, for broadband and sufficiently
stationary sounds considered from a frequency domain viewpoint, pre- and post-echoes as found
in the equalization filter matrix are masked by the stationary nature of the sounds under test.
Loudness, spectrogram and power spectal densities (PSD) were computed using a
commercial software. The recorded microphone signal was filtered by high-pass filters with
cut-off frequency at 20 Hz. Loudness versus time was computed using FFT according to
standard ISO 532 B with: Hanning window, FFT size of 16384 samples, overlap of 75 %. The
FIGURE 1: Typical aircraft cabin sound. (a): Sound pressure level versus time, (b) Loudness versus time, (c) Power
spectral density versus time and (d) Average power spectral density from a 4-second sample.
PSD and PSD versus time were computed on the basis of 16384-sample frames, Hanning
window and 75 % overlap.
SIGNAL PROCESSING: MULTICHANNEL SOUND FIELD REPRODUCTION
The signal processing is represented in Fig. 2 as a block diagram in the frequency domain. A
microphone array was used for the capture and evaluation of the target and reproduced sound
fields, respectively. Reproduction sources inside the cabin are used for sound field reproduction.
ˆ
G
#
[k] G[k],
ˆ
G[k]

+
M L M
ˆ p s
M
ˆ pe
−jω∆
e
−jω∆
r, ˆ r
E[k],
ˆ
E[k]
ˆ e
FIGURE 2: Block diagram of the sound field reproduction system and signals.
The plant including the reproduction sources, the vibroacoustic responses of the mock-up
and the microphone array is described for each frequency bin k =0, 1, ..., N−1 by a complex
matrix G[k] ∈ C
M×L
where M is the number of microphones, L the number of reproduction
sources for the frequency bin k under consideration and N is the number of points in the
frequency domain. The measurement of G will be denoted
ˆ
G (measured quantities are indicated
by ˆ· ). The multichannel equalization filter is denoted by a complex matrix
ˆ
G
#
[k] ∈ C
L×M
. This
filter transforms, by matrix multiplication, the M measured and digitized target signals in the
frequency domain ˆ p[k] ∈ C
M
into L digital transducer signals s[k] ∈ C
L
. This is written as follows
s =
ˆ
G
#
ˆ p. (1)
Ideally, the combination of the equalization filter with the actual plant should lead to a white
and decoupled system E[k] with a modeling delay of ∆ samples E=G
ˆ
G
#
≈Ie
−jω∆
, with
0 ≤ω≤2π, the normalized angular frequency, and ω=2πk/N. Matrix E∈ C
M×M
represents the
equalized system. The modeling delay is introduced to ensure that the equalization filters
ˆ
G
#
are causal. Any deviation of the equalized system E from the identity matrix with additional
modeling delay e
−jω∆
will lead to reproduction errors.
The measured target sound field at the microphone array is denoted by ˆ p∈ C
M
and r[k] ∈ C
M
is the reproduced sound field at the microphone array with r =Eˆ p=Gs and s =
ˆ
G
#
ˆ p. The
measurement of the reproduced sound field r in the cabin mock-up will be denoted ˆ r. An ideal
physical reconstruction of the sound field would lead to ˆ r =r = ˆ pe
−jω∆
. A useful physical
quantity is the reproduction error vector at the microphone array ˆ e[k] ∈ C
M
= ˆ pe
−jω∆
− ˆ r.
One way to circumvent the limitation of the brute-force inversion approach to the
aforementioned problem (
ˆ
G
#
=
ˆ
G
+
e
−jω∆
, with
+
denoting pseudo-inversion) is to include a
regularization of the solution norm in a least-mean-square reproduction error minimization.
The solution is defined by (Gauthier et al., 2012)
s =argmin{
ˆ
Gs−e
−jω∆
ˆ p
2
2

2
s
2
2
} (2)
where the amount of regularization is controlled by the penalization parameter λ. Vector
2-norm is denoted by ·
2
. We assume that λ is fixed for all frequencies. The solution of Eq. (2)
is given by
s =(
ˆ
G
H
ˆ
G+λ
2
I)
−1
ˆ
G
H
e
−jω∆
ˆ p, (3)
and the equalization matrix is
ˆ
G
#
=(
ˆ
G
H
ˆ
G+λ
2
I)
−1
ˆ
G
H
e
−jω∆
. (4)
The penalization enhances the main diagonal of the denominator matrix, hence preventing the
instability of the inversion and reducing the solution 2-norm s
2
2
.
EXPERIMENTAL SETUP: MICROPHONE AND TRANSDUCER ARRAYS
The target sound fields have been recorded by a planar microphone array in single 2-hour
flight aboard a CRJ900 jet aircraft in various conditions. This microphone array was described
in an earlier paper (Gauthier et al., 2012). The microphone array is made of 80 electret
microphones and preamplifiers. The array installed in the aircraft cabin for the actual in-flight
recording is shown in Fig. 3. The array configuration is based on a uniform rectangular grid
array with additional variation over two layers along the vertical axis to distinguish propagating
directions in the vertical direction. The approximate microphone-to-microphone distance is
12.5 cm so that spatial aliasing should be avoided up to ≈1.4 kHz at the microphone array.
FIGURE 3: In-flight recording with 80-channel microphone array (back), binaural manikin (right, front), 32-channel
spherical microphone array (right, back), reference microphones (not shown) and accelerometers (not shown).
The cabin mock-up is shown in Figs. 4(a) and 5(a). It models a Bombardier CRJ900 with 8
seats and 6 windows. Mock-up details are given in (Gauthier et al., 2012). The main
reproduction transducers are 32-mm inertial actuators mounted on internal trim panels. Four of
these actuators are mounted on each of the 9 trim panels. To induce vibration in the floor, 4 bass
shakers are attached to the wood floor. A subwoofer is included. Actuators positions are shown
in Fig. 4(b). The microphone array inside the cabin mock-up is shown in Figs. 5(a) and (b). The
microphone array is identical to the one used for the in-flight recording (see Fig. 3). In the
mock-up, the microphone array has two purposes: 1) identification of the plant
ˆ
G and 2)
measurements of the reproduced sound fields.
(a)
0
1
2
3
−1
−0.5
0
0.5
1
−1
−0.5
0
0.5
1
x
2
[m] x
1
[m]
x
3

[
m
]
(b)
FIGURE 4: (a) Exterior view of the mock-up. In this paper, the external loudspeakers are not used. (b) Geometrical
arrangement of the trim-panel and floor actuators inside the mock-up.
(a)
x
2
x
1
x
3
(b)
FIGURE 5: (a) Microphone array in the mock-up. (b) Configuration of the 80-channel microphone array.
The 3280 (41 actuators × 80 microphones) frequency response functions of the plant matrix
ˆ
G were measured with the logarithmic swept sine method using 200 averages for the trim panel
actuators and 400 for the floor shakers and the subwoofer. The amplitudes of the swept sines
were adjusted to avoid audible rattles or non-linear distortions caused by the trim panels. The
resulting system was stored in matrices that include all these FRFs and IRs with zero-padding
up to 24 000 samples (0.5 seconds). The modeling delay ∆ was set to 12 000 samples.
The 4 floor shakers and subwoofer are in charge of reproducing the low frequency range of
the target sound field at 4 of the microphones while the 36 trim-panel shakers must reproduce
the remainder of the audio bandwidth over the 80 microphones. For most of our previous
experiments, the crossover frequency was 220 Hz and was achieved using simple low-pass and
high-pass 4th-order Butterworth filters (Gauthier et al., 2012). We used forward and reverse
filtering to ensure that the phases of the crossover filters are linear. This also gives a -6 dB gain
at the cut-off frequency so that there is no amplification at the crossover frequency once the
low-frequency and high-frequency range systems are combined. In this paper, we investigate a
finer definition of an optimal crossover frequency that provides a better transition from the low
to the high frequency ranges using similar high-pass 4th-order Butterworth filters.
EXPERIMENTAL RESULTS: OBJECTIVE EVALUATION
To quantify sound field reproduction at the microphone array in the frequency domain,
several metrics are introduced. These metrics are defined in the discrete frequency domain k
where signal spectra are represented by one-sided modified periodograms obtained using
Welch’s method and Hanning window. For the reported experiments, the modified Welch
periodograms were evaluated with a Fourier transform size of 8192 samples, an overlap of 50 %
and a Hanning window. The averaged and normalized reproduction error is
E
LS
[k] =
ˆ e[k]
2
ˆ p[k]
2
, (5)
it describes, on the average, how accurately the sound field is reproduced over the microphone
array. E
LS
=0 means a perfect reproduction. To quantify the reproduced sound environment
with respect to timbre, the averaged magnitude spectrum error is defined in dB ref 1 / Hz by
M
LS
[k] =10log
10
| ˆ p[k]| −10log
10
| ˆ r[k]|
2
/M, (6)
it represents the reproduction error in terms of power density function of the measured ˆ p[k] and
reproduced ˆ r[k] sound environments, i.e. it does not take into account the spatial distribution of
the phase but only the spatial distribution of sound pressure amplitude. In Eq. (6), absolute
value | · | and log
10
are elementwise. The source signal 2-norm is given by
S
LS
[k] =s[k]
2
. (7)
In this paper, two parametric studies are reported: effect of regularization parameter λ and
crossover frequency. As mentioned in (Gauthier et al., 2012), the bass management was initially
introduced to circumvent the presence of squeaks and rattles in the trim panels if driven by
strong low frequency content. However, in this paper, the test sounds derived from real in-flight
recording implied significant squeak and rattle noises in reproduction experiments. Therefore,
for the parametric studies and objective evaluations of the signal architecture reported in this
paper, the original microphone array signals were artificially reduced in amplitude to minimize
any squeaks and rattles. This highlights the fact that sound field reproduction inside aircraft
cabin mock-ups using vibration transducers should rely on high-quality materials and careful
assembly of the trim panels. These considerations should be addressed at the design stage of the
mock-up regrouping engineers and technicians in charge of mock-up fabrication.
Results: Effect of regularization
From theory and Eq. (2), it is expected that a smaller λ should give smaller reproduction
errors. True in theory, it is not exactly true in practice: a small regularization parameter λ
increases the reproduction source signals and may lead to a large reproduction error if the
digital-to-analog converters saturate or if squeaks and rattles in the trim panels introduce
non-linear distortions. Here we investigate the effect of λ on the objective performance of
reproduced sound field at the microphone array. Results for the low-frequency system are shown
in Fig. 6(a) while results for the high-frequency system are shown in Fig. 6(b). We conduct
separate studies for the two frequency ranges since non-linear distortions in one frequency
range might degrade the sound field reproduction in the other range. The best results, in terms
of E
LS
and M
LS
, are obtained with λ=0.01 and λ=0.1 for the low and high frequencies,
respectively. For the high-frequency range, it is interesting to note the sudden increase in E
LS
and M
LS
for the smallest regularization parameter (λ=0.01), this typically corresponds to
emerging squeaks and rattles. From Fig. 6(b), one can also note that on the average the M
LS
metric is below 1 dB/Hz and is relatively flat for the broadband noise. Fig. 7 shows the
corresponding power spectral densities.
10
2
10
3
10
−3
10
−1
10
1
E
LS
E
L
S


10
2
10
3
0
1
2
3
M
LS
M
L
S

[
d
B

r
e
f

1

/

H
z
]


0.001
0.004
0.007
0.01
0.001
0.004
0.007
0.01
10
2
10
3
10
−10
10
−5
10
0
S
LS
S
L
S
Frequency [Hz]


0.001
0.004
0.007
0.01
(a)
10
2
10
3
10
4
10
−3
10
−1
10
1
E
LS
E
L
S


10
2
10
3
10
4
0
1
2
3
M
LS
M
L
S

[
d
B

r
e
f

1

/

H
z
]


10
2
10
3
10
4
10
−10
10
−5
10
0
S
LS
S
L
S
Frequency [Hz]


0.01
0.04
0.07
0.1
0.01
0.04
0.07
0.1
0.01
0.04
0.07
0.1
(b)
FIGURE 6: (a) E
LS
, M
LS
and S
LS
for various λ. The crossover frequency was set to 220 Hz and is shown as a thick
dashed vertical line. (a): Low frequency range, (b): High frequency range.
10
2
10
3




Freq. [Hz]
P
o
w
e
r

/

F
r
e
q
.

[
d
B

r
e
f

1

/

H
z
]
)
Welch Power Spectral Density Estimate at Mic #14, y−grid: ∆ 20 dB


Target
0.001
0.004
0.007
0.01
(a)
10
2
10
3
10
4




Freq. [Hz]
P
o
w
e
r

/

F
r
e
q
.

[
d
B

r
e
f

1

/

H
z
]
)
Welch Power Spectral Density Estimate at Mic #14, y−grid: ∆ 20 dB


Target
0.01
0.04
0.07
0.1
(b)
FIGURE 7: Example of power spectral density of the target and reproduced sound fields for microphone #14. (a): Low
frequency range, (b): High frequency range.
Results: Effect of crossover frequency
In this section, we look for a better crossover frequency to improve the transition from the
low- to the high-frequency range, corresponding to a change of the plant dimension (from 5
transducers and 4 microphones in the low-frequency range to 36 transducers and 80
microphones in the high-frequency range). Results obtained when changing the crossover
frequency in the equalization matrix obtained with optimal λs as identified earlier are shown in
Fig 8. The best crossover frequency is 140 Hz. Using this new crossover frequency and best
regularization parameters, an example of power spectral density at a single microphone is
shown in Fig. 9. Clearly, there is a good agreement between the target and reproduced sound.
10
2
10
3
10
−3
10
−1
10
1
(a)
E
L
S


10
2
10
3
0
0.5
1
(b)
M
L
S

[
d
B

r
e
f

1

/

H
z
]


10
2
10
3
10
−10
10
−5
10
0
(c)
S
L
S
Frequency [Hz]


140 Hz
160 Hz
180 Hz
200 Hz
220 Hz
240 Hz
140 Hz
160 Hz
180 Hz
200 Hz
220 Hz
240 Hz
140 Hz
160 Hz
180 Hz
200 Hz
220 Hz
240 Hz
FIGURE 8: (a) E
LS
, M
LS
and S
LS
for various λ for varying crossover frequency (frequency axis is zoomed).
10
2
10
3
10
4





Freq. [Hz]
P
o
w
e
r

/

F
r
e
q
.

[
d
B

r
e
f

1

/

H
z
]
)
Welch Power Spectral Density Estimate at Mic #14, y−grid: ∆ 20 dB


Target
Reproduced
FIGURE 9: Power spectral density of the target and reproduced sound fields for microphone #14 for the best reported
case, λ=0.1 for the low-frequency range, λ=0.01 for the high-frequency range and the crossover frequency is 140 Hz
(thick dashed line).
DISCUSSION AND CONCLUSION
It was shown that sound field reproduction at a microphone array is possible for real in-flight
sound recordings using standard multichannel leat-mean-square equalization. In all cases, the
squeaks and rattles issue is not a limitation of the proposed signal processing but of our specific
mock-up materials. Results are satisfactory if one takes into account several limitations and
compromises: 1) the use of an overdetermined problem (M>L) and regularization to prevent
excessively large reproduction signals that could lead to audible squeaks and rattles from the
trim panels and 2) the need for a satisfactory global sound reproduction in the mock-up. Indeed,
for the second issue, using a smaller microphone arrays with M=L could give very low
reproduction error at the microphone array but with larger reproduction errors outside the
microphone array, a consequence that is not acceptable. Current and future works are oriented
towards to the adaptation of the improvement least-mean-square method.
ACKNOWLEDGMENTS
This work is part of a project involving: Consortium for Research and Innovation in
Aerospace in Québec, Bombardier Aéronautique and CAE, supported by a Natural Sciences and
Engineering Research Council of Canada grant.
REFERENCES
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field reproduction inside aircraft cabin mock-up”, in Proceedings of the 133rd Audio
Engineering Society Convention.
Janssens, K., Vecchio, A., and der Auweraer, H. V. (2008). “Synthesis and sound quality
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