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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

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

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

81

82

ZVEREV et al.

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

ACOUSTICAL PHYSICS

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2009

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)

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)

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)

should sum the mode signals in view of their amplitudes and phases. This procedure is an inversed one

ACOUSTICAL PHYSICS

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83

modes of the ideal waveguide:

PA2O ( , v )

=

(7)

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)

84

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-

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

Vol. 55

No. 1

2009

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,

ACOUSTICAL PHYSICS

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2009

85

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.

ACOUSTICAL PHYSICS

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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.

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

ACOUSTICAL PHYSICS

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2009

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-

88

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.

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

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.

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

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

ACOUSTICAL PHYSICS

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89

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.

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

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)

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.

REFERENCES

1. M. Fink, D. Cassereau, A. Derode, et al., Rep. Prog.

Phys. 63, 1933 (2000).

2. V. A. Zverev, Akust. Zh. 50, 685 (2004) [Acoust. Phys.

50, 685 (2004)].

3. W. S. Hodgkiss, H. C. Song, W. A. Kuperman, et al.,

J. Acoust. Soc. Am. 105, 1597 (1999).

4. S. Kim, W. A. Kuperman, W. S. Hodgkiss, et al.,

J. Acoust. Soc. Am. 115, 1525 (2004).

5. H. C. Song, S. Kim, W. S. Hodgkiss, and W. A. Kuperman, J. Acoust. Soc. Am. 116, 762 (2004).

6. H. C. Song, W. S. Hodgkiss, and W. A. Kuperman,

J. Acoust. Soc. Am. 118, 1381 (2005).

7. C. Prada, Julien de Rosny, D. Clorennec, et al., J. Acoust.

Soc. Am. 122, 761 (2007).

ACOUSTICAL PHYSICS

Vol. 55

No. 1

2009

8. K. G. Sabra, P. Roux, H.-C. Song, et al., J. Acoust. Soc.

Am. 120, 1305 (2006).

91

11. B. G. Katsnelson and V. G. Petnikov, Acoustics of Shallow Sea (Nauka, Moscow, 1997) [in Russian].

(2006) [Acoust. Phys. 52, 180 (2006)].

14. V. A. Zverev, A. A. Stromkov, and A. I. Khilko, Akust.

Zh. 52, 676 (2006) [Acoust. Phys. 52, 784 (2006)].

15. V. A. Zverev, Radio Optics (Sov. Radio, Moscow, 1975)

[in Russian].

16. V. A. Zverev, Physical Fundamentals of Imaging by

Wave Fields (Institute of Applied Physics, Ross. Akad.

Nauk, Nizhni Novgorod, 1998) [in Russian].

Zh. 51, 221 (2005) [Acoust. Phys. 51, 175 (2005)].

Translated by E. Kopyl

Zh. 54, 69 (2008) [Acoust. Phys. 54, 58 (2008)].

10. V. A. Zverev, P. I. Korotin, and A. A. Stromkov, Akust.

Zh. 54, 637 (2008) [Acoust. Phys. 54, 552 (2008)].

ACOUSTICAL PHYSICS

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2009

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