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ISSN 1063-7710, Acoustical Physics, 2009, Vol. 55, No. 1, pp. 8191. Pleiades Publishing, Ltd., 2009.

Original Russian Text V.A. Zverev, P.I. Korotin, A.A. Stromkov, 2009, published in Akusticheski Zhurnal, 2009, Vol. 55, No. 1, pp. 6273.

OCEAN ACOUSTICS
AND UNDERWATER SOUND

Modal Time Reversal of Waves for Shallow-Water Sea


V. A. Zverev, P. I. Korotin, and A. A. Stromkov
Institute of Applied Physics, Russian Academy of Sciences,
ul. Ulyanova 46, Nizhni Novgorod, 603950 Russia
e-mail: zverev@hydro.appl.sci-nnov.ru
Received November 28, 2007

AbstractA method is proposed for calculating the time-reversed wave field generated by a point source in a
waveguide by using signals received by a vertical antenna array. The procedure of time reversal is based on representing the wave field in the form of the decomposition into modes of the ideal waveguide. In contrast to the
earlier proposed simplified numerical method of time reversal of waves, the method presented here allows one
to obtain the reversed field for the entire thickness of the waveguide. The method is successfully applied to a
shallow-water sea with a depth of 120 m, at distances of 7, 10.5, and 12 km. It is shown that an opportunity
arises to increase the gain of the array; to determine the parameters of the medium, including its stability in the
presence of currents; and to match the point of transmission of an arbitrary received signal and the point of
transmission of the reversed signal.
PACS numbers: 43.60.Tj, 43.30.Vh, 43.30.Wi, 43.39.Bp
DOI: 10.1134/S1063771009010096

INTRODUCTION
Time reversal of waves (TRW) [1, 2] allows one to
focus the transmitted signal passing through a composite scattering and dispersing medium. Such a technique
has been successfully used in the shallow-water sea [3
10]. The focusing is performed with the use of two vertical antenna arrays. Let them be denoted as A1 and A2.
The A1 array is used for receiving the signal to be
reversed and then for transmitting the time-reversed
received signal. The A2 array serves to transmit the
reversed signal from one of its elements and to receive
the reversed signal transmitted by the A1 array into the
medium for determining the vertical distribution of the
reversed field. It was proposed in [9] to substantially
simplify the procedure of determining the shape of the
focal spot formed in the medium by time reversal of the
wave. This proposal consists in using a single vertical
array, A1, and a single point sound source instead of the
A2 array.
It is shown in [9] that the field calculated with the
use of that technique coincides with the reversed field
at only a single point. However, that point is the center
of the reversed field: here, the maximum of the reversed
field exists. As for the rest of the points, the field decays
at them, just as the reversed field does. There are no
other similarities between the calculated and reversed
fields. One cannot calculate the full reversed field by
the method proposed in [9] if the signal to be reversed
is transmitted from a single point.
The calculation of the reversed wave requires knowing the frequency response (FR) of the medium
between all the points of the A1 and A2 arrays. However, the use of a single transmitting point allows one to

determine the FR of the medium between the single


point from which the reversed signal is transmitted and
all the points of the A1 array that received that signal.
Thus, if the A1 and A2 arrays consist of N and M elements, respectively, one should know N*M frequency
responses of the medium while only N of them are
known from the experiment. Such considerations show
that the calculation of the reversed field with the use of
a single receiving array without transmitting the signal
into the medium is fully impracticable.
The aforementioned situation is characteristic of the
general case of an arbitrary dispersing and scattering
medium. However, it is not so in the case of sound propagation in a shallow sea with sufficiently large number
of modes that can propagate in that waveguide but with
a small number of the modes actually propagating in it.
In such a situation, those contracted data allow one to
obtain the required information on the FR of the
medium between all the points of the A1 array and the
points of the medium that compose a virtual A2 array,
that is, the points at which the reversed field is calculated over the entire thickness of the waveguide. The
objective of our work is to solve that problem on the
basis of an experiment performed in a shallow-water
sea [9, 10, 1214].
The signals propagate in the form of modes in a
shallow-water sea. Each mode is represented by a wave
propagating in the entire waveguide. At the same time,
each mode propagates with its own frequency response
(MFR) of the waveguide. In the approximation at hand,
a simple and universal relation exists between the MFR
of a separate mode and the FR between individual
points of the medium. By decomposing the field
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ZVEREV et al.

received by the A1 array into individual modes, one can


determine the MFR of the waveguide for those modes
and then estimate the FR between all the points of the
A1 and A2 arrays. Accurately solving the problem is
hampered by the fact that the thus-calculated MFRs
carry information on amplitudes and phases of the
modes excited by the sound source. Usually, these characteristics of the signal and the source are well known.
However, in a number of cases, one can make use of
signals transmitted by unknown sources. Then, the
unknown parameters of the signal and the source hinder
determination of the depth of the source of the signal to
be reversed, but, in contrast to [9], they allow one to calculate the shape of the focal spot formed in time reversing the field over the entire thickness of the waveguide.
Here, we study this possibility for an experiment
performed in a shallow-water sea as an example. Earlier, such experimental data were used in the works [9,
10, 1214]. In contrast to the method of time reversal
with transmission of the time-reversed signal into the
medium, the proposed technique is denominated as the
modal reversal of waves (MRW) because the main
instrument of the technique consists in decomposing
the direct signal into modes of the waveguide, modifying the excitation factors, and subsequently synthesizing a new signal. Such a process does not require transmitting the reversed signal into the medium.
It will be shown that one can choose an arbitrary
depth of the source of the signal to be reversed, in
accordance with experimental needs. In doing so, one
achieves higher noise immunity in signal reception and
obtains information on the specificities of sound propagation on the path, not only for the point at which the
signal to be reversed was transmitted but also for the
chosen depth. The MFR will be shown to deeply focus
both the signal to be reversed, whose shape is known,
and any other signal emerging from the point corresponding to the signal to be reversed. Signals from
other points and noises are focused only slightly. Note
that the neither of the features of the given procedure
can be obtained by using the classic TRW technique
with transmission of the signal into the medium.
1. FREQUENCY RESPONSE
OF THE WAVEGUIDE
FOR AN INDIVIDUAL MODE
To decompose the vertical distribution of the wave
field into the signals of individual modes and to subsequently sum them after the procedure of time reversal,
one should know the shape of the signal at a certain
mode along the vertical, that is, the mode profile. We
specify such a dependence in the form known for an
ideal waveguide [11]. For a chosen frequency band, this
dependence can be approximated by the following formula [10, 11]:

UN ( n, m ) = sin ---- nm .
N

(1)

Here, n is the ordinal number of the element of the vertical array covering the entire water layer (if the array is
not insufficiently long, signals of the missing elements
are specified to be zero); m is the ordinal number of the
mode; and N is the number of elements in the array covering the entire layer. In our case, N coincides with the
number of all propagating modes for the central frequency of the band [11]. The mode number m is an integer only in the case of a perfectly soft bottom [11]. For
the bottom that has some impedance, the mode number
is not an integer. The bottom impedance is unknown.
Therefore, we use the approach proposed in [1214] to
calculate Eq. (1). Quantity m is not specified as an integer but it is defined in the form of a continuous series of
numbers with an arbitrary step (0.2 in our case), up to
some upper limit.
Actually, in a real waveguide, the form of the mode
profile can differ from that of Eq. (1). Nevertheless, one
can represent the real field in the waveguide as an
expansion in functions corresponding to Eq. (1). For
such an expansion to be efficient, it is necessary and
sufficient that the obtained series should show good
convergence. To that end, the field must be excited by a
point source, and the mode shapes of the natural
waveguide must not significantly differ from Eq. (1). It
is shown in [1214] that the expansion in functions of
Eq. (1), which represents the field in the waveguide at
hand, converges well. According to [13], the angular
spectrum of individual terms in the decomposition onto
modes of the ideal waveguide coincides well with the
theoretically calculated angular spectrum of the signals
corresponding to individual modes of the ideal
waveguide. Such a coincidence can be explained by the
fact that, for our experiment, the sound speed is nearly
independent of depth (the vertical distribution of the
sound speed is presented in [13] for that experiment)
and the bottom is sufficiently plain and uniform.
The factors of the expansion in functions (1) were
calculated in accordance with the usual mathematical
procedure that, in the case of the ideal waveguide, has
the following form for the Fourier spectra of the modes
[13, 14]:
MA1 ( , m ) =

UN ( n, m )PA1 ( , n ),

(2)

where PA1(, n) are complex Fourier spectra of the signals received by the nth element of the A1 array.
Let us introduce function ZM (, m) which represents the MFR for the mth individual mode of the
waveguide. The Fourier spectrum of the signal for the
mth mode selected by the A1 array can be expressed in
terms of the MFR in the following way [10]:
MA1 ( , m ) = ZM ( , m )UN ( j, m )G ( ).

(3)

Here, j is the ordinal number of the element of the virtual A2 array. That element transmits the reversed signal with the Fourier spectrum G(). In the expression
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MODAL TIME REVERSAL OF WAVES FOR SHALLOW-WATER SEA

for MA1, number j is omitted because the corresponding information cannot be used in reversing the signal.
Thus, if Eq. (3) is divided by the spectrum of the transmitted signal, one will obtain the MFR of the mth mode
with the accuracy up to the excitation factor for the corresponding mode, UN( j, m). That factor is redundant
and even undesired for us because its appearance in the
time-reversed field leads to the loss of information on
the depth at which the signal to be reversed was transmitted. That depth can be arbitrarily chosen in the time
reversal procedure.
2. TIME REVERSAL OF WAVES IN MODES
OF IDEAL WAVEGUIDE
Until now, we have used the considerations published in [9, 10, 1214]. Now, let us make an important
step. Let us consider a virtual A2 array that has a sound
source as one of its elements. By using Eq. (3), one can
perform reversal of waves in the signals of modes of the
ideal waveguide, that is, decompose the field into those
signals at the virtual A2 array. The spectra of the mode
signals at the A2 array are obtained by multiplying the
spectra of the mode signals (3) by the complex-conjugated MFR. However, the latter cannot be explicitly
determined from the data of our experiment. Instead of
the MFR, we use the following function that is completely determined by the known signals:
F()
ZMO ( w, m ) = MA1 ( , m ) -------------,
G()

(4)

where F() is the function that is equal to 1 within the


frequency band of the signal and 0 out of that band. By
substituting Eq. (3) into Eq. (4), we obtain the expression that replaces the MFR in the experiment:
ZMO ( , m ) = ZM ( , m )UN ( j, m )F ( ).

(5)

For the time reversal, we use only the phase of the


experimentally obtained MFR but not its modulus.
Thereby, we get rid of the complications caused by the
irregularities of the MFR under the influence of interference. In the ideal waveguide, the modulus of the
MFR is equal to unity, and only its phase works. This
proves that the phase of the MFR is sufficient for
reversing the signal in the natural waveguide as well.
By using Eq. (5), the signal determined from the experiment in the form of modes of the waveguide at the
A2 array can be written as follows:
ZMO* ( , m )
MA2O ( , m ) = MA1 ( , m ) --------------------------------- .
ZMO ( , m )

(6)

To obtain the reversed signal at the A2 array, one


should sum the mode signals in view of their amplitudes and phases. This procedure is an inversed one
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with respect to that used in decomposing the signal into


modes of the ideal waveguide:
PA2O ( , v )
=

MA2O ( , m )UN ( v , m )UN ( q, m ).

(7)

Factor UN(q, m) is introduced into Eq. (7) to provide


for focusing the reversed signal at the chosen virtual
hydrophone with number q in the A2 array. Parameter q
can take on any value except for those making the introduced factor to be equal to zero.
In view of Eqs. (3) and (5),
PA2O ( , v ) =

ZM* ( , m )ZM ( , m )

----------------------------------------------------ZM ( , m )
m

(8)

( UN ( j, m ) )
------------------------------UN ( , m )UN ( q, m )G ( ).
UN ( j, m )
2

Equation (8) shows that the thus-performed calculations solve the main problem; namely, this expression
really reverses the wave field because it contains the
product of the actual MFR of the waveguide and the
complex-conjugated expression.
Let us now consider to what extent the reversal of
the wave field according to Eq. (8) corresponds to the
TRW result. To do so, let us perform a mental experiment. Assume that we can transmit the signals received
by the A1 array in the time-reversed form and receive
those signals by the real A2 array. In such a mental
experiment, the Fourier spectrum of the signal at the
A2 array is given by the following formula [10]:
PA2 ( , v , j ) =

ZM ( )UN ( , m )ZM* ( )
m

(9)

UN ( j, m )G* ( ).
Factor ZM() appearing in Eq. (9) describes signal
propagation from A1 to A2 in the form of modes (it
makes no difference in which direction the signal propagates [4]). According to Eq. (3), the last three multipliers represent the time-reversed Fourier spectrum of the
signal at A1 in the form of the decomposition into
modes of the waveguide. Factor UN(, m) and summing over m serve to inversely transform the signal represented as the waveguide modes into the Fourier spectrum of the wave field.
The field reversed with the use of transmitting the
reversed signal into the medium also contains a member with the product of the waveguide MFR and the
complex-conjugated function. Comparing Eqs. (8) and
(9) shows that the dependence on parameter j in Eq. (8)
is not quite the same as in Eq. (9), which provides for
focusing the reversed field at the place where the source
of the signal to be reversed exists. However, Eqs. (7)

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ZVEREV et al.
(a)

50

(b)

(c)

(d)

50
0

100

100

50

100
(e)

150

5
10 0 10 10 0 10 10 0 10
10
(f)
(g)
(h)
15

10

10

20
5

20

40

60

80

10 0 10 10 0 10 10 0

10

Fig. 1. Upper row: signals received by A1 at a distance of (a) 10.5 km and by A2 at distances of (b) 7, (c) 10.5, and (d) 12 km. Lower
row: the result of decomposing the field into modes of the ideal waveguide at A1, at a distance of (e) 10.5 km, and at A2, at distances
of (f) 7, (g) 10.5, and (h) 12 km. Time (in ms) is laid along the horizontal axis. In vertical: (ad) sea depth (in m) and (eh) mode
numbers. The range of brightness is 20 dB.

and (8) are added with such a member that the reversed
signal would be focused at the place chosen by us.
3. DATA OF THE IN-SEA EXPERIMENT
In the experiment, the vertical receiving array consisting of 32 hydrophones equidistantly spaced with a
step of 3 m over a 93-m length was bottom-moored at a
depth of 120 m. The array was fixed and self-contained.
Signals from all the array elements were multi-channelly memorized. Pulsed broadband (100300 Hz) linearly frequency-modulated (LFM) signals with a duration of about 5 s were transmitted from a vessel drifting
with an embedded sound source. Thus, it was possible
to obtain signals transmitted from different distances.
The thickness H of the water layer was greater than the
length of the array. For this reason, signals of the eight
upper elements of the array were replaced by zeros.
The experimentally obtained data are shown in the
figures. Figure 1 presents the key features of the exper-

iment on time reversal of the waves with transmission


of short pulses. In the experiment, long LFM signals
were transmitted into the medium, rather than the
aforementioned short pulses. A short pulse with the unit
spectrum in the entire frequency band of the signal
received by the A1 array, 100 to 300 Hz, was formed in
signal processing. To do so, spectra of the LFM signals
received by the A1 array were divided by the complex
spectrum of the transmitted LFM pulse, G(), within
the frequency band of the received signals.
Figure 1 has the following structure. The left-hand
side of the figure shows the signals at the A1 array: the
received signals and the same ones represented in the
form of modes according to Eq. (2) are presented at the
top and bottom of that part of the figure, respectively.
The right-hand side shows the reversed signals at the
A2 array, whose spectra are calculated according to
Eq. (8). Below, the same signals are presented in the
form of reversed modes at A2 whose spectra are calculated with the use of Eq. (6). The figure shows the modACOUSTICAL PHYSICS

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MODAL TIME REVERSAL OF WAVES FOR SHALLOW-WATER SEA

uli of the signals represented in an analytic form. The


real signals are converted into complex ones by a transformation analogous to that used in [15, 16] for transforming the real wave field. In practice, the procedure
is as follows: within one half of the frequency band
(either one), the spectrum is artificially specified to be
zero. After that, the signal obtained by inversely Fourier
transforming becomes complex, with the imaginary
and real parts coupled by the Hilbert transformation
with each other. The real part of the obtained complex
signal is the signal itself, and the modulus of the complex signal yields its amplitude. Thus, all the calculated
signals are presented in the form of moduli in the figure.
Such a representation allows one to exclude the carrier
frequency from the signal, as it carries no information
on its amplitude and only shows the position of its spectrum on the frequency axis.
The calculations were performed in the following
order. From the signals whose moduli are shown in
Fig. 1a, the signals of modes of the waveguide were
computed (Fig. 1e). Then, the time-reversal of the
mode signals was performed (Figs. 1f and 1g). After
that, the mode signals were used to calculate the field at
the A2 array (Figs. 1b1d).
Figure 1a shows moduli of the signals received by
the A1 array upon transmission of a short pulse with a
duration of 5 ms corresponding to the 200-Hz width of
its spectrum. It can be seen that the received signals
have a much greater duration than the transmitted
pulse. This phenomenon is caused by the dispersion of
the velocity of wave propagation in the waveguide and
by reverberation. The expected value of the dispersion
can be estimated by using a formula for the group delay
of the waves presented in [11]. According to that formula, the group delay is 13 ms and almost 1 ms at frequencies of 100 and 300 Hz, respectively, for the first
mode of the waveguide. For the third mode, the delay at
100 Hz will be higher by an order of magnitude,
namely, 117 ms, while it will remain the same, 13 ms,
at 300 Hz. These estimates are quite sufficient to
explain the blooming of the multimode signal in the
waveguide, which can be seen from Fig. 1a. Similar
results were presented in [3] for the data obtained in the
Mediterranean Sea with comparable parameters of the
experiment.
Figures 1b1d show the signals received from all
three distances and calculated for the virtual A2 array
(the coordinates are the same: sea depth and time). In
these figures, the signal is compressed in time down to
the duration of the initial short pulse, and it is substantially compressed in depth as well. The compression of
the signal is determined by the number of modes propagating in the waveguide. For calculating the signals
shown in Figs. 1b1d, twelve modes of the ideal
waveguide are used in decomposing the field into
modes. Increasing the number of modes does not lead
to further narrowing of the obtained focal spot in depth,
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85

while decreasing of this number yields increased spot


size.
According to Fig. 1e, as few as six modes of the
ideal waveguide are sufficient to represent the field at
the A1 array. This is just what was done in the initial
version of the paper. Figures 1f and 1g illustrate the
decomposition of the field into signals of the same
modes but with reversal of these signals at the A2 array.
Here, one can clearly see that six modes are insufficient, and all twelve members of the decomposition
must be used. The reason for this is that the higher
modes suffer from very strong dispersion. The signals
of these modes blur, and they cannot be seen at the A1
array. Reversal of the mode signals fully eliminates
their blurring caused by both the dispersion of the propagation velocity in the waveguide and the reverberation. The mode signals are compressed, increase, and
become visible. The data of Figs. 1f1g show that the
used decomposition of the field into modes of the ideal
waveguide converges quite well.
Thanks to the use of the short pulse as the transmitted signal, the data shown in Fig. 1 can be compared
with the published data of the experiments performed
using the TRW technique [3]. The time-reversed pulses
shown in Fig. 1 differ only slightly from the similar
reversed signals obtained in the Mediterranean Sea [3]
at the same depths and the same distances as in our
experiment.
The result of reversing the total signal at the virtual
A2 array is shown in Fig. 2 for two distances of 7 and
12 km used in the experiment. For the same distances,
the pattern of the reversed modes of the signal at A2 is
shown in Fig. 3. The LFM signal is characteristic in that
the frequency changes proportionally to time in it:
instead of time, the horizontal axis in Fig. 3 may be
measured in frequencies. At the beginning, low frequencies on the order of 100 Hz arrive; then, the frequencies increase to 250 Hz. Figure 3 shows the bend
in the curves of the moduli of the mode signals with the
change in the signal frequency. This phenomenon can
be explained by a small change in the bottom parameters accompanying the change in the frequency on the
path at hand.
4. ESTIMATION OF QUALITY AND NOISE
IMMUNITY OF SIGNAL TIME-REVERSAL
The signal received by the A1 array can be processed by three different methods. One can simply sum
the signals received by all the elements of the array:
K (t) =

A1 ( t, n ),

(10)

where A1(t, n) is the signal received by the nth element of the A1 array at time t. Let us denote this
method as K.

86

ZVEREV et al.
(a)

50
0
100
5
0

(b)

10
15

50

20

100
0

Fig. 2. Moduli of the signal obtained by modal time reversal at distances of (a) 7 and (b) 12 km. The scale of brightness is logarithmic, with a range of 20 dB. Time (in s) and sea depth (in m) are laid along horizontal and vertical axes, respectively.

(a)
10

5
0
0
0

(b)

(b)

5
10

10
15
20

Fig. 3. Moduli of the mode signals of the waveguide obtained by modal time reversal at distances of (a) 7 and (b) 12 km. The scale
of brightness is logarithmic, with a range of 20 dB. Time (in s) and sea depth (in m) are laid along horizontal and vertical axes,
respectively.
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MODAL TIME REVERSAL OF WAVES FOR SHALLOW-WATER SEA

87

5000

5000

1.25

1.30

1.35

1.40

Fig. 4. Fragment of oscillograms of the signals received by the A1 array at a distance of 12 km and processed by the methods F, S,
and K (upper, middle, and lower plots, respectively). Time (in s) and the oscillograms on a linear scale are shown in horizontal and
vertical, respectively.

1.0

10

0.8

0.6

20

0.4

30

K
5

10

40
50
60
0.10 0.08 0.06 0.04 0.02

0.02

0.04

0.06

0.08

Fig. 5. Coefficients of correlation between the signals received by the A1 array at a distance of 12 km and the initial signal processed
by the methods F, S, and K (solid line, circles, and crosses, respectively). Time (in s) and the level of the correlation coefficient (in
dB) are laid along horizontal and vertical axes, respectively. Maximal values of the correlation coefficient are shown in the box for
all three processing methods.

In coherently summing, one can also account and


compensate for the influence of the frequency
responses of the medium, as in [9, 10]:
PS ( j, ) =

ZS* ( )g ( , n ).

(11)

j, n

Here, PS(j, ) is the Fourier spectrum of the thus-processed signal, ZSj, n() is the frequency response of the
medium between the point j of the signal transmission
and the nth element of the receiving A1 array, and g(, n)
is the Fourier spectrum of the signal received by the nth
element of the A1 array:
g ( , n ) = ZS j, n ( )G ( ).

(12)

Frequency response ZSj, n() of the medium is determined with use of Eq. (10) [9, 10] by dividing the complex Fourier spectrum of the signal received by each
nth element of the A1 array by the complex Fourier
spectrum of the signal transmitted into the medium. Let
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us denote the method of signal processing according to


Eq. (11) as S.
Finally, one can use the result of the modal signal
reversal (8) and take the signal from the chosen depth q
where it is maximal. Let us denote this method as F.
Figure 4 shows a fragment of the oscillogram of the signal received by the A1 array from a distance of 12 km
and processed by all three methods. The figure shows a
great difference between the oscillograms obtained by
the modal reversal of waves (F) and all other methods.
The oscillogram has the shape of a perfect sinusoidal
signal that is not distorted by interfering noises. At the
same time, the signals processed by the S and K procedures are substantially distorted and in some places
exhibit fractures, rather than appearing as perfect sinusoids.
Figure 5 shows the correlation coefficients of the
signals correlated with the initial signal in processing
by the K, S, and F methods. The magnitude and the
duration of the correlation coefficient in time are objec-

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ZVEREV et al.
Q = 0 dB

Q = 32.003 dB
0

0
5

300

300

10
15

200

200

20

12

25

100

100
16

30

20

35
0

Fig. 6. Moduli of current spectra of the signals received at a distance of 12 km and processed by the K method. Time (in s) and
frequency (in Hz) are laid along horizontal and vertical aces, respectively. The current spectrum within a dynamic range of 36 dB
is shown at the left. At the right, the same spectrum with eliminated signals higher than 32 dB is presented. Quantity Q characterizes the level of the maximal signal relative to the initial one which is shown at the left.

tive characteristics of the quality of time reversal: the


higher and narrower the correlation coefficient, the
closer the correlated signal to the initial one and the
higher the quality of reversal. According to the figure,
the highest quality of reversal corresponds to the
F method, that is, to the reversal of the signal in the
form of modes. The high quality of reversal does not
fully guarantee the highest noise immunity because the
immunity depends on two factors, namely, the ways in
which both the signal and the noise transform.
To determine the extent of the noise immunity of the
modal time reversal, the signals and the noise were
extracted from the same time realization. To do so, time
filtering of the sliding spectrum of the realization was
used. This procedure is described in the Appendix.
Figure 6 shows the sliding spectrum used in extracting the signal, noise, and additional signals S1 and S2.
The figure presents the current spectrum of the signal
received from a distance of 12 km and simply pro45
40
35
30
25
20

Fig. 7. Signal-to-noise ratio for the signals processed by the


methods F, S, and K (crosses, squares, and solid line,
respectively). Time (in s) and the signal level (in dB relative
to the noise) are laid along horizontal and vertical axes,
respectively.

cessed by the K method. According to the Appendix,


processing in this way allows one to detect signals differing in their levels or in the shapes of the current spectra. Thus, all signals whose level was higher than 32 dB
relative to the maximum were eliminated from the signal shown in the left-hand part of Fig. 6. The result is
the signal whose current spectrum is shown at the right;
only noise remains in this signal. A similar signal was
obtained for each element of the A1 array, then processed by all three methods and used for estimating the
signal-to-noise ratio.
Figure 7 shows the time dependence of the signalto-noise ratio for the unreversed signal K and the
reversed signals S and F. Both reversed signals exceed
the unreversed one in their noise immunity but there is
no signal with the highest noise immunity among them.
At low frequencies, the S processing has an apparent
advantage, while the F method is the best at high frequencies. The noise immunity of the reversed signals is
the result of their higher coherence in comparison with
the noise at the receivers of the A1 array. The noise
immunity of the reversed signals can be treated as an
increase (or a decrease) in the gain of the array.
Questions arise as to whether the noise is focused at
the chosen point and how the individual signals that can
be extracted from the current spectrum are focused. To
answer these questions, two more signals named S1 and
S2 were extracted from the current spectrum shown at
the left in Fig. 6. The result of focusing all the extracted
signals and the noise is shown in Fig. 8. According to
this figure, the noise is focused much more weakly than
the main signal. Signal S1, extracted separately, is
focused to the same extent as the main signal. This fact
indicates that the S1 signal emanates from the same
point as the main one. It seems that the intense generation of the main signal is accompanied by the generation of the second harmonic due to the resonance of the
transmitter, the harmonic emanating from the same
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0
10
20
30
40
50

20

40

60

80

100

120

Fig. 8. Focusing of the signals extracted from the current


spectrum according to the method described in the Appendix. The main LFM signal, S1 and S2 signals, and noise are
shown by dotted, long-dashed, short-dashed, and dash-anddot lines, respectively. The signal level (in dB relative the
main focused LFM signal) and the sea depth are laid along
the vertical and horizontal axes, respectively.

sound source into the medium with the same FR. The
aforementioned example shows the possibility of using
the focused signal for determining the position of the
sound source relative to the position of the reversed signal. Signal S2 is focused only slightly. This fact means
that the S2 signal is transmitted from a point other than
the point of transmission of the main signal. This example shows that the point from which a signal emanates
can be determined relative to the point from which the
testing signal was transmitted.
CONCLUSIONS
Let us consider some conclusions of the study performed. It seems that, for calculating the reversed field
without transmitting the signal into the medium, one
requires much information on the medium because of
its complexity. This is especially true for the case of a
signal propagating in a shallow-water sea, at a distance
of 12 km. It seems that large number of surveys of the
medium must be carried out, and then the obtained data
should be accounted for in a complex computer code.
In fact, there is no need for all those procedures and one
does not require any information on the medium. The
only thing that is needed is the shape of the signal transmitted into the medium. This information offers the
opportunity to obtain the complex FR of the medium,
which accounts for all the features of signal propagation, including both the dispersion and various time
delays. With the use of the obtained MFR for the ideal
waveguide, one can focus the signal and eliminate the
changes introduced in it by sound propagation in the
medium. In some cases, one needs no a priori information at all. This situation takes place when signals that
are usually used for studying the FRs of different
devices and media are transmitted into the medium. In
particular, such signals include short individual pulses
and linearly frequency-modulated signals that were
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used in our studies. In these studies, the transmitted signal was not known a priori because the frequency of
transmitting was not thoroughly fixed but formed automatically.
Knowing the shape of the transmitted signal, one
can obtain the FR of the medium, study it, and get some
features of signal propagation, which cannot be
obtained in other equally simple and reliable ways.
Because of the compensation for the distortions
caused by the medium, the time reversal in a shallowwater sea increases the noise immunity of signal detection. Moreover, the proposed method offers additional
opportunities for studying the features of sound propagation on the path. For instance, Fig. 3 exhibits a bend
in the curves of modes as a function of the current frequency of the signal. Such a bend can be treated as a
sign of change in the bottom parameters along the path
for different signal frequencies. Note that the classical
TRW procedure with transmitting signals of A1 into the
medium cannot be used for achieving an additional
noise immunity, and there is no way to obtain any additional information on the medium (only the compensation takes place). The term modal reversal of waves
emphasizes the applicability of the method only to
waveguides, in contrast to the general TRW method
that can be applied to arbitrary media.
The method proposed and considered here allows
one to study the stability of the medium as well. In that
task, the modal time reversal differs only slightly from
the classic TRW but exceeds the latter in the possibility
of studying the media with any currents, in contrast to
the TRW. To study the stability of the medium, the signal received by the A1 array at a certain time should be
processed with the use of the signals received by the
same array at other times, with subsequent observation
of time differences in the reversed signals. The medium
is stable in the case when there are no significant differences in the time-reversed signals.
ACKNOWLEDGMENTS
This work was supported by Russian Foundation for
Basic Research, projects nos. 08-02-00818 and 07-0201205.
APPENDIX
DETECTING SIGNALS WITH DYNAMIC FILTER
An important characteristic of the noise immunity
of an antenna array with the use of different methods of
signal processing is the signal-to-noise ratio. To determine that characteristic on the basis of a time realization in which signal and noise exist simultaneously, one
should extract the signal and noise separately and then
calculate their ratio. The specificity of the LFM signal
used by us is that such a signal has a narrow spectrum
within each sufficiently short time interval while the
noise has a wide spectrum in an arbitrary time interval.

90

ZVEREV et al.
(a)

(b)

(c)

(d)

(e)

(f)

300
200
100
0
300
200
100
0

Fig. 9. FF filters and the signals extracted by them: (a) filter of the LFM signal and (d) signal extracted by it; (b) filter of the S2 signal
and (e) the signal itself; (c) filter of the S1 signal and (f) the signal itself. Time nT (in s) and the frequency (in Hz) are laid along
horizontal and vertical axes, respectively.

We used this specificity of the LFM signal to separate


the noise from the signal in their mutual time realization.
For this reason, the initial signals were represented
in the form of gliding spectra. To do so, the time interval (about 5 s) occupied by the signal was divided into
30 parts T, 128 intervals of time sampling per part, and
then spectrally analyzed within each time interval T:
( n + 1 )T

F ( nT , ) =

f ( t ) exp ( it ) dt.

(A.1)

nT

Here, n is the ordinal number of interval T and T is the


time interval that, in our case, was 1/30 of the duration
of the realization in which signal and noise exist simultaneously. In this way, Fig. 6 was obtained, in which
time nT is laid along the horizontal axis and frequency
is shown in vertical, up to half of the sampling frequency.
To filter such a signal, we constructed the filtering
function (FF) in the form of a dependence in the same
coordinates, Q(nT, ). The current spectrum can be
used to obtain the FF. In this case, the FF is equal to
unity when F(nt, ) reaches values higher than a certain
threshold and to zero at the other case. With such a FF,
the transformed current spectrum takes the following
form:
FP ( nT , ) = F ( nt, )Q ( nt, ).

(A.2)

Thus, the current spectrum of the noise shown in


Fig. 6 at the right was obtained from the current spectrum shown at the left. The procedures of Eqs. (A.1)
and (A.2) are reversible because the Fourier transform
is reversible as well. For obtaining the time realization

of the signal filtered with the use of the FF, it is sufficient to perform the following transformation:
0.5

fp ( t ) =

n 0.5

d
FP ( nT , ) exp ( it ) ------- ,
2

(A.3)

where is the sampling frequency. Here, each individual summand represents the signal starting at time t =
nT and terminating at time t = (n + 1)T. Summation over
n in Eq. (A.3) means combining all the intervals, each
of which has duration T, into a mutual interval with the
duration of the entire realization.
Function FF can be formed in another way. The signals that have certain positions in the current spectrum
represented in coordinates nT, can be extracted (or
eliminated) from this spectrum.
Figure 9 shows examples of functions FF and the
signal extracted with the use of them, namely, the main
LFM signal and the S1 and S2 ones.
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