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Apparatus for detecting composite broad-band frequency-modulated signals with filtration in scale-time domain based on discrete wavelet transform |
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IPC classes for russian patent Apparatus for detecting composite broad-band frequency-modulated signals with filtration in scale-time domain based on discrete wavelet transform (RU 2439601):
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Coordinate system is created from GIB buoys with base length of 1-3 km, which receives ping signals of underwater objects, synchronised with GPS clocks and time-spaced. Through correlation reception, the GIB buoys determine lag time from each object and relay these data to a control station. Based on the lag time and data on hydrostatic pressure on the underwater object, the control station calculates coordinates and displays the position of each object.
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Proposed device represents adaptive system to allow optimising antenna phase reception of acoustic signals in Fresnel range. For this, proposed device comprises multi-component cylindrical antenna with N receiving channels. It differs from known designs in that it incorporates additionally HF signal generator and HF radiator and allows every preamplifier to switch over to frequency multiplexer mode. Said distinctive features allow heterodyning received useful HF signal and optimising antenna phase reception of acoustic signals with curved wavefront.
![]() Method involves generation and emission from a source of a harmonic signal with frequency ω, reception of an acoustic signal using a set of N≥8 hydrophones which form a circular measuring base directed into the horizontal plane, picking up quadrature components of the complex envelope of received acoustic signals, measurement of the phase of acoustic signals, preliminary phasing of the measuring base into N directions passing through the centre of the measuring base and each of the N hydrophones, determination of the direction which corresponds to the maximum of the signal and a hydrophone lying in that direction, calculation of the heading angle to the source in a local coordinate system associated with the measuring base using corresponding formulas. The hydrophone lying in the direction of the signal maximum is taken as the first hydrophone. The mobile subsurface object is also fitted with a pair of hydrophones spaced out in a diametrical plane along the mobile subsurface object at a distance of 1≤λ/4σθ. After calculating the bearing, the heading angle β0 to the source is calculated using formula β0=β1±θ0, σβ0=σβ, where β1 is the heading angle of the first hydrophone of the circular measuring base, the sign (+) is taken for the heading angle of the starboard side, the sign (-) is taken for the heading angle of the port side, σβ0 is the error in determining the heading angle, σθ is the bearing measurement error. The mobile subsurface object then synthesises a beam path on which the condition β0=180°+σθ is satisified, and a traversing path on which the condition β0=±90°+σθ is satisfied. Further, phase difference of acoustic signals Fm received using an extra pair of hydrophones is measured on the traversing path at time moments tm, m=1-M. The values of phase difference of acoustic signals Fm(tm) measured on the acoustic path are approximated with a linear function F=a(t-t0). Parametres a are determined through a least-squares method using corresponding formulas, and the true heading angle to the source at point t=t0 is determined using formula
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Invention is related to the field of hydroacoustics, namely to devices for detection of narrow-band noise hydroacoustic signals (with spectral density of power in the form of separate discrete components or their scales) at the background of additive noise. Invention is based on calculation of continuous wavelet transformation of input process on the basis of complex analytical wavelet, relative band of amplitude spectrum of which matches relative band of spectral density of detected signal power. Device comprises analog-digital converter (ADC) 1, recirculator 2, the first calculator of fast Fourier transform (FFT) 3, complex multipliers 4.1 - 4.M, scaling devices 5.1 - 5.M, device of complex conjugation 6, device of negative frequencies nulling 7, the second calculator FFT 8, permanent memory (PM) 9, calculators of reverse FFT 10.1 - 10.M, calculator of module square 11, averaging device 12, threshold device 13, control device 14.
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Invention is related to hydroacoustics and may be used for protection of objects from the barrier side in water medium. According to method, signal is generated from hydroacoustic antenna arranged in the form of piezoelectric cable sections, ends of which are connected to radio frequency cable with the help of matching devices fed from common source, signal voltage is picked up from loading resistor and is sent through separating capacitor to inlet of alarm signal generator, object parametres are identified by results of analysis of spectral and time variation of signal.
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Present invention can be used for determining the trajectory of a supersonic projectile. At least the initial part of signals is measured, containing information only on impact wave, using five or more acoustic sensors, spread out in space such that they form an antenna. From this measured initial part of signals, the difference in arrival time for a pair of sensors is determined. A genetic algorithm is applied to the initial chromosome, which contains initial estimated parameters of the projectile trajectory. For a given number of generations, projection errors are calculated for solutions, obtained from chromosomes from the genetic algorithm. The ratio of solution with the least values of projection errors to the ambiguous solution is calculated, and if this ratio is greater than a given value, the solution with the least value of calculated projection error is chosen as the correct trajectory of the projectile.
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Noise signals are received in horizontal and vertical plane, frequency-time processing is carried out in every spatial channel of observation, output voltages of formed space channels are squared and summed in all frequency samples, then averaged in time, signals are centered and normalized to noise, signal energy and information parameters are accompanied, route detection is carried out by comparison of generalised weight of signal local maximums with threshold of signal detection, which corresponds to threshold ratio of signal-noise. Method is based on the fact that in every cycle of viewing noise signals are received, primarily processed, squared, secondarily processed and route-detected in at least another two frequency ranges and additionally, at least, for two angles of observation in vertical plane, creating new expanded set of space channels.
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Determined is the starting point for autonomous underwater robots (AUR), taken for the beginning coordinates. Control ship in moved in accordance with the movement of the AUR. Onboard of the AUR the coordinates are determined, which are then controlled by the base hydro-acoustic beacon, on which is additionally added a transmitter of navigational signals which emits navigational signals. Navigational signals are received onboard the AUR, processed and combined with the information signal. Evaluation of the AUR coordinates are obtained by the data of the hydro-acoustic navigation system (HANS), which is made complex, and a deliberate evaluation is made of the coordinates AUR. This data is transmitted with AUR by the hydro-acoustic channel, the base hydro-acoustic beacon is set, then transmitted through a cable link to onboard the control ship and is reflected in real time.
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Method includes as follows. Horizontal and vertical orientation characteristic static fan receives noise signals in combination with frequency-time processing within each spatial observation channel, quadrating, time averaging, alignment and signal normalising to interference, observation of current view cycle for received normalised signals and detection decision-making comparing to limit value of signal-interference relation. Thus within each view cycle for each frequency sample the adaptive spatial observation channels are formed, at least by three adjacent spatial channels in horizontal or vertical plane.
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Invention concerns television systems for underwater inspection. The arm contains underwater research equipment with photographic and video equipment mounted thereon, connected to picture monitor on control panel and supplied with electrically driven lifting gear. The arm is provided with flat arrow-shaped steel wing front-located with three vertical stabilisers serving as construction supporting foot. The wing is cable-towed through lifting gear by water vehicle. Transmitter of surveying echosounder is placed with direction response pattern on the bottom side vertically coaxial with the receiver of satellite grid station. Emitting sector contains control unit, electric motor case with headed screw and two bars fixing provisional weight attached to wing. Two guides between bars are furnished with sealed boxes and underwater lamps provided on both sides. View areas of photographic and video equipment established in sealed boxes are mutually crossed within surveyed surface. The whole view area of photographic and video equipment is overlapped with illumination sectors and two acoustic signal transmitters detecting wing plane position relative to surveyed surface. Real-time control, management and data transfer is performed through multicore cable connecting control unit, picture monitor and operator's stand.
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Direction finder can be used for taking azimuth relatively guarded objects at guarded areas, calculating number of objects in group target and classifying found objects. Direction finder has two seismic receivers, two processing channels with delay lines and correlators, maximal signal selector, correlator, testing module, commutator and calculator. To realize the direction finding function the method of passive diversity detection and ranging is used. The main information criterion for finding direction to object has to be the function of mutual signals correlation in two signal processing channels. Value of azimuth is judged from value of signal delay. Change in value of signal delay is equivalent to controlling directional diagram of seismic active aerial which allows classifying detected objects separately. Test influence is used for adaptation of speed of propagation of seismic wave which changes under influence of meteorological conditions. Current value of speed of propagation of seismic wave is judged from time of delay of test influence signal coming to second seismic receiver. Tuning of lines of delay is conducted correspondingly to those changes.
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The method includes reception of the signal of noise radiation of the noisy object by the first receiving antenna and spectral analysis of the received signal of noise radiation of the noisy object, reception of the signal of noise radiation is also performed by the second receiving antenna, separated is the reciprocal spectrum of the signals of noise radiation received by the first and second receiving antennas, measured is the value of the carrier frequency of the autocorrelation function, and the decision on the class of the noisy object is taken at comparison of the measured carrier frequency of the autocorrelation function with threshold frequencies, each being determined as an average frequency of the initial noise radiation band of each standard object of a definite class.
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Method includes determining, in the moment of temporary position of expanding spatial angles wave front, tracking belonging to acoustic beam (bearings) for each reflective element, positioned in wave packet of reflected signal (in space between frontal and back fronts of signal pulse, and limited body angle of direction characteristic of receiving antenna. Spatial receipt on basis of spatial-phase and spatial-correlative processing of reflected signal provides for detecting difference between spatial positions of reflecting objects within received signal wave front, which provides more information for object detection and, due to that, principally distinguishes the method from commonplace amplitude-temporal signals processing technology.
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In the method, receipt of acoustic signals is performed by two linear groups of sound receivers. In first and second processing channels, electric signals are processed at frequency f, received by first and second linear groups of sound receivers, and in channel of frequency f1 - signals with frequency f1, received by first one of linear groups of sound receivers. Bearing to sound source is determined with utilization of relation of voltage amplitudes at outputs of first and second processing channels. Amplitude of signal voltage at output of first processing channel is connected, with supposition, that sound source is positioned on working axis of normalized characteristic of direction of first one of linear groups of sound receivers. Amplitude of sound pressure at input of first one of linear groups of sound receivers at frequency f is formed by dividing calculated value on proportionality coefficient, determined experimentally at frequency f. Level of sound pressure is calculated at input of first one of linear groups of sound receivers. Analogical calculations are performed for signal at frequency f1. Type of substrate surface is determined, and decrease of sound pressure level, caused by influence from obstructions, meteorological and atmospheric factors. Distance and topographic coordinates are calculated with consideration of influence of aforementioned factors.
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In the method, receipt of acoustic signals is performed by two linear groups of sound receivers. In first and second processing channels, electric signals are processed at frequency f, received by first and second linear groups of sound receivers, and in channel of frequency f1 - signals with frequency f1, received by first one of linear groups of sound receivers. Bearing to sound source is determined with utilization of relation of voltage amplitudes at outputs of first and second processing channels. Amplitude of signal voltage at output of first processing channel is connected, with supposition, that sound source is positioned on working axis of normalized characteristic of direction of first one of linear groups of sound receivers. Amplitude of sound pressure at input of first one of linear groups of sound receivers at frequency f is formed by dividing calculated value on proportionality coefficient, determined experimentally at frequency f. Level of sound pressure is calculated at input of first one of linear groups of sound receivers. Analogical calculations are performed for signal at frequency f1. Type of substrate surface is determined, and decrease of sound pressure level, caused by influence from obstructions, meteorological and atmospheric factors. Distance and topographic coordinates are calculated with consideration of influence of aforementioned factors.
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In device for determining direction to a source of sound, consisting of two photo-electric shadow devices and information processing systems, laser beams are directed at an angle of 90° to each other. In each photo-electric shadow device after focusing objective laser beam is split onto two laser beams, and these two laser beams go to two knives with mutually perpendicular edges. Edge of one of aforementioned knives in each photo-electric shadow device is parallel to plane, parallel to laser beams. Information, received from two photo-receivers, standing behind these knives, is utilized for maintaining similar sensitivity of both photo-electric shadow devices. Output signals from one of these photo-receivers and two other photo-receivers of photo-electric shadow devices are squared, amplified and added. Signal at output of adder is maintained constant due to loop of negative check connection from output of adder to inputs of amplifiers. On basis of signals at outputs of amplifiers with consideration of mutual phases of signal at outputs of photo-detectors by means of phase detectors and electronic computing machine, direction towards sound source is determined.
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Noise direction finder comprises three vector receivers whose directional characteristics are oriented along the Cartesian co-ordinate system, amplifiers, band filters, three-channel unit for processing information, and computer.
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In accordance to method, recording of sound signals is enabled in case of registration of impact waves from by-flying ultrasound bullet and barrel wave from expanding gases from barrel edge by sensitive elements, processing of these signals by means of processor, on basis of results of which position of sound source is determined. Method contains following innovations: sensitive elements are preliminarily fastened immovably relatively to optical axis of video recording device, synchronously with recording of sound signal by not less than 3 sensitive elements, recording of video image of possible position of sound source is performed by means of at least one video recording device, mounted with possible change of filming direction and position in space, during following processing of signals moment of arrival of barrel wave and frame from recorded video row, closest to aforementioned moment, are combined, and mark of rifleman position is placed on that frame. Recording of video image is performed in optical or infrasound or other range.
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Method for using navigational hydro-acoustic system by underwater devices includes determining position of leading underwater device relatively to responder beacons on basis of distances to responder beacons, determined by measuring expansion times of acoustic signal from underwater device to responder beacons and back. Position of each following underwater device is determined on basis of difference of total distances from leading underwater device to each responder beacon and from each responder beacon to following underwater device and distance from leading underwater device to following underwater device, determined by measuring onboard the following underwater device of differences between moments of receipt of acoustics signals of request of responder beacons by leading underwater device and responses of responder beacons, and distance to leading underwater device and direction towards it, known onboard the following autonomous underwater device.
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Mode of using by underwater vehicles of a navigational hydro acoustic system is in simultaneous determination of the locations of all underwater vehicles of the group at inquiry by a hydro acoustic signal-command of one of the underwater vehicles of the group of (leading) responder beacons by one of the (driven) responder beacons. The location of each of underwater vehicles is determined by differences of distances to the leading responder beacon and to the drive responder beacon defined by measured intervals of time between reception of an acoustic signal of the request of the responder beacons by the leading responder beacon and acoustic signals of the response of the driven responder beacons. The location of the underwater vehicle is found as an intersection plot of hyperboloid of revolution whose number corresponds to the number of pairs of "leading-driven" responder beacons and focal points are located in installation plots of the corresponding responder beacons and the flatness passing through the center of the hydro acoustic antenna of the underwater vehicle transversely to the flatness of the true horizon.
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FIELD: physics. SUBSTANCE: apparatus has an analogue-to-digital converter (1) whose output is connected to the input of a recirculator (2). The output of the recirculator is connected to the first input of a discrete wavelet transform computer (3). The wavelet transform output is connected to the input of a fast Fourier transform computer (8), the output of which is connected to the first input of a complex multiplier (9), the output of which is connected to the input of an inverse fast Fourier transform computer (11), the output of which is connected to the input of a squared absolute value computer (12), the output of which is connected to the input of a threshold device (13), the output of which is the output of the apparatus; a control device (14), the outputs of which are connected to control inputs of the analogue-to-digital converter (1), the recirculator (2), the discrete wavelet transform computer (3), a multiplier (5), an inverse discrete wavelet transform computer (7), the fast Fourier transform computer (8), the complex multiplier (9), the inverse fast Fourier transform computer (11) and read-only memory (4, 6, 10). EFFECT: reduced computational and hardware expenses when filtering a signal in a scale-time domain. 15 dwg
The invention relates to the field of sonar and radar systems, namely the complex detectors broadband frequency-modulated signals of known shape on the background of additive noise. It is known that the implementation of best practice in solving the problem of detection signals against the background noise is largely determined by the level of knowledge about the incoming signal. For signals with unknown initial phase is optimal quadrature receiver [2-4] (analog), providing a slight loss (1-1 .2 dB) compared with a matched filter. The main disadvantage of quadrature reception is limited its application only to the class of narrow-band signals. In the case of broadband signals necessary multichannel scheme, performing quadrature filtering for each component. If the phase spectrum of the signal is unknown, use energy methods and devices receiving [2-4] (analog), which is a sequential execution of the filtering operations, quadratic detection and integration. The drawback of such methods is the effect of the suppression of small signal", which is a consequence of the fact that the output signal-to-noise ratio (BSA) is proportional to the square of the input PCB. If the form of the received signal s(t) is completely known is on (except for the amplitude A and the time of arrival of the signal t 0: As(t-t0)), the potential robustness in solving the problem of the detection signals (including broadband frequency modulation) on a background of white noise, in principle, provides a correlation receiver or coherent filter [1 (s-343), 2-4] (analog). Moreover, the robustness of these two methods of detecting a known signal correlator and matched filter) in a theoretical sense exactly the same. The generalized signal-to-noise ratio (BSA) at the output of these detectors, defined as the ratio of the increment component of the output of the process is the mathematical expectation E[y(t)], due to the presence of the useful signal, the variance of the output process in the absence of a signal D[y0(t)], equal to twice the signal energy Es(taking into account the random amplitude) to the spectral density of the noise N [1] Data discovery methods known signal implicitly assume one of the types of one-dimensional representation of signal and noise: either in the spectral region (matched filtering), or in the time domain (correlation method). However, for the input process x(t) sonar or radar receiver includes a two-dimensional representation - scale-time domain, obtained by p is the so-called imeneniya the wavelet transform (EAP) [5-9]. Continuous EAP can be defined as the scalar product of the investigated process x(t) and the basic wavelet function ψατ(t) [5] where the hell are the top denotes the complex conjugation operation. The General principle of construction of the basis of the EAP is to use large-scale transformations with parameter compression ratio α and shifts with the shift parameter τ of the source wavelet function ψ(t) (the so-called mother wavelet) To be a wavelet, the basis functions ψατ(t)∈L2(R) should have the required properties [5-9]. They should be quadratically integrable, alternating (and have zero mean), the wavelets must tend to zero at The formula for continuous inverse wavelet transform has the form As can be seen from (4), the original signal x(t) can be recovered through the integral sum of the same basis functions ψατ(t) with weights in view of the wavelet spectrum of the signal [W ψx](α,τ). Here the constant Cψ(3) acts as a normalizing factor, similar to the factor (2π)1/2, normalizing the Fourier transform. For better calculation operators direct (1) and reverse (4) EAP can be represented in the frequency domain [5, 9]. When this is achieved a substantial improvement in the performance of digital devices that implement the EAP, by calculating convolutions using effective procedures FFT. The operator (1) continuous direct EAP can be defined in the frequency domain [5 (p.59, 67-68), 9] where
Operator (4) continuous reverse EAP can also be defined in the frequency domain as where The only limitation to this form of account operators (5) and (6) continuous EAP, compared with (1) and (4), is the requirement of analyticity for the studied signal and the applied wavelet [5 (p.67-68)] ie You know, large numbers the creation of other types of time-frequency representation of signals (Tabor, Page, Wigner, Cswa-Williams, etc.) [10], but they all (unlike EAP) have the worst confining properties in the time-frequency plane (have a false items and bad simultaneous appearance of Delta-pulse and tone) and do not always have the inverse transformation for exact recovery of the original time signal. Accordingly, these methods of time-frequency representation is usually only used for signal analysis, but not for solving problems of detection on the background noise. The use of scale-temporal representation (based on IP) to the input process x(t) of the detector signals [11] where s(t) is detected echoes; n(t) is the additive interference as Gaussian white noise, allows (up directly matched filtering) to implement preliminary "bandpass filtering the wavelet spectrum of the received signal, s(t) simultaneously in the field of time and scale (frequency) using a special filter H(α, τ) where as a scale-time filter H(α, τ) is a two-dimensional function of a special form where A0- the level specified on the basis of the conditions option is 0<B<1 is selected depending on the type of frequency modulation signal and the storage conditions specified amount of energy of the original signal Pre-filters (8) of the input process x(t) in the scale-time domain (for the most widely used in sonar and radar class of frequency-modulated signals) allows to achieve a significant increase in noise, compared to the classical coherent filter or correlation receiver [1-4]. Output CPEs proposed detector rightly be written in the form: The gain in noise immunity of the proposed method of detection CMS compared with the classical coherent filtering caused by the reduction of the noise level (N1<N) after wavelet filtering" (8) the input process, while maintaining approximately constant the energy of the received signal Reducing the spectral density of the noise (as well as the e variance) and accordingly, the increase in GSP (characterizing the gain in noise immunity) is approximately equal to the ratio of "squares" scale-temporal region occupied by the wavelet spectrum of the input process to "wavelet filtering" Wx and after wavelet filtering" Wx1 as in the case of white noise, its power (after VP) is uniformly distributed on a large scale the time domain within a conditional "rectangle", limited on the time axis τ - pulse duration of TSand along the axis of the scale of the α - band magnitude ΔαS=α1α2(clearly the corresponding spectral band of the frequency deviation of the modulated signal ΔfS=fin-fn). After filtration (8) remaining in the output process x1(t) of the power interference scale-time domain is limited to the area of media filtration function (α, τ)∈supp{[H(α, τ)}, where H(α, τ)≠0. The method of detection of frequency-modulated signals filtered scale-time domain [11] includes the following operations. 1. The calculation of the wavelet transform Wx(α,τ) of the input process x(t) (the most effective this procedure is implemented in the frequency domain using analytic wavelet, in accordance with the operator (5)): 1.1. the choice of the source wavelet, ψ(t) computation of its range For the 1.2. the calculation of the basis spectra analytic wavelet by scaling (compression) of the original spectrum of the mother wavelet: 1.3. the calculation of the Fourier spectrum of the input process 1.4. the multiplication of Fourier spectrum of the input process 1.5. calculating the inverse Fourier transform on the result of the last multiplication: 2. The calculation of the wavelet transform Ws(α,τ) of the reference probe signal s(t) (using the previously calculated of the basis spectra analytic wavelets 2.1 calculation of the Fourier spectrum of the reference signal 2.2 the multiplication of Fourier spectrum of the reference signal 2.3 in the overall number of General inverse Fourier transform on the result of the last multiplication: 3. The formation of two-dimensional filter function H(α,τ) scale-temporal plane, the clip region of values of (α,τ), where the magnitude of a complex wavelet spectrum of the standard probing signal Ws(α,τ) exceeds a certain level, A0(13): 4. The multiplication of the wavelet spectrum of the input process and scale-temporal filtering functions:
5. The calculation of the inverse wavelet transform for the last multiplication Wx1(α,τ) (using the previously calculated of the basis spectra analytic wavelets 5.1 calculation of the direct Fourier transform in τ from the two-dimensional wavelet spectrum 5.2 multiplication of Fourier spectrum 5.3 calculation of the inverse Fourier transform by f from the last multiplication: 5.4 integration 5.5 calculation of the real part of the restored process Further processing of the process x1(t) coincide with the classical implementation of the method detection signal of known shape-based matched filtering. 6. Matched filtering of the detected signal s(t) known form: 6.1. calculation of complex Fourier spectrum 6.2. calculation of complex Fourier spectrum 6.3. the multiplication of the complex Fourier spectrum 6.4. calculating the inverse Fourier transform on the result of the last multiplication: 7. Selection (quadratic detection) of the envelope of the response of the matched filter: (calculating, key writing, the square module 8. Comparison It should be noted that when using the envelope of the response SF Note also that the operations 1.1 and 1.2 are made only with the wavelet ψ(t), operation 2.1, 2.2, 2.3, 3, and 6.2 are made only with the reference signal s(t)and not with the test input process x(t), and thus, these operations can be carried out in advance, and the results of their calculations is stored in ROM. The device [11] (prototype)that implements a method for detecting the complex of broadband frequency-modulated signals filtered scale-time domain [11], is shown in figure 1, where: unit 1 - analog-to-digital Converter (ADC); block 2 - recirculator; unit 3 - the solver fast Fourier transform (FFT) 1; blocks 4.1-M - complex multiplier products; block 5 - permanent memory (ROM) 1; blocks 6.1-M - solvers inverse FFT; unit 7 - matrix complex is the multiplier; unit 8 - ROM 2; blocks 9.1-M - FFT solvers; the blocks 10.1-M - complex multiplier products; unit 11.1-M - solvers inverse FFT; block 12 - matrix integrator; unit 13 - the transmitter 2 FFT; unit 14 - the complex multiplier; block 15 - ROM 3; block 16 - calculator inverse FFT; block 17 - transmitter unit square; block 18 - threshold device; block 19 - control device. Thus, the detection unit complex broadband frequency-modulated signals filtered scale-temporal region (prototype) [11] includes: analog-to-digital Converter (unit 1), the input of which is applied the input signal, the ADC output is connected to the input of the recirculator (block 2), the output of which is connected to the input of the first transmitter fast Fourier transform (block 3), the output of which is connected with the first inputs of the M complex multiplier products (blocks 4.1-M)whose outputs are connected to inputs of M computing the inverse Fourier transform (units 6.1-M)the outputs are connected to first inputs of a matrix of complex multiplier (block 7), the outputs of which are connected with inputs M computing the fast Fourier transform (blocks 9.1-M), the outputs of which are connected with the first inputs of the M complex multiplier products (blocks 10.1-M)whose outputs are connected to inputs of M will calculate the MDL inverse fast Fourier transform (blocks 11.1-M), the outputs are connected to inputs of matrix integrator (block 12), the output of which is connected to the input of the second transmitter fast Fourier transform (block 13), the output of which is connected to the first input of complex multiplier (block 14), the output of which is connected to the input of the transmitter inverse fast Fourier transform (block 16), the output of which is connected to the input of the transmitter unit square (block 17), the output of which is connected to the input of the threshold device (block 18), the output of which is an output device; the first persistent storage device (block 5), the outputs of which are connected with the second the inputs of the M complex multiplier products (blocks 4.1-M) and with the second inputs of the M complex multiplier products (blocks 10.1-M); the second persistent storage device (block 8), the output of which is connected with the second input matrix complex multiplier (block 7); the third permanent storage device (block 15), the output of which is connected to a second input of the complex multiplier (block 14); control unit (unit 19), the outputs of which are connected to control inputs of analog-to-digital Converter (unit 1), recycler (block 2), calculators fast conversion Fourier (blocks 3, 9.1-M and 13), the complex multiplier products (blocks 4.1-M, 7, 10.1-M and 14), computing the inverse fast Fourier transform (block the 6.1-M, 11.1-M and 16) and permanent storage devices (blocks 5, 8, and 15). The principle of the device is as follows. To the input device enters the implementation of the input process x(t), which is fed to the input of the ADC (block 1) with a sampling rate that satisfies the requirements of the sampling theorem: ADC output (block 1) discrete samples are sent to the input of the recirculator (block 2), which is formed and with each new count is updated to the current discrete sample x(n) of length N samples. Sample length N is determined by the duration of the emitted signal and the sampling interval: With outputs of the first read-only memory (block 5) reads M one-dimensional arrays of length N samples (pre-calculated basis With the output of the inverse FFT solvers (units 6.1-M) the result of the wavelet transform of the input process Wx(m, n) in the form of a two-dimensional array of size M of N scale shifts is supplied to the first inputs of the matrix complex multiplier (block 7). From the output of the second ROM (block 8) is read pre-calculated two-dimensional array of filter function H(m, n) (scale-temporal plane) of size M N scale shifts and is supplied to the second input matrix complex multiplier (block 7), the outputs of which M one-dimensional arrays of length N samples of the multiplication (i.e. filtering scale-temporal plane) Wx1(m, n) are fed to the inputs of the FFT solvers (blocks 9.1-M), which outputs the results of the FFT With outputs of the first read-only memory (block 5) reads M one-dimensional arrays of length N samples (pre-calculated basis Output matrix integrator (block 12) temporary implementation of the filtered scale-temporal process area x1(n) is fed to the input of the second transmitter FFT (block 13), the output of which a comprehensive range From the third ROM (block 15) is read pre-calculated complex conjugate spectrum of the reference signal The control device (block 19) synchronizes operation: analog-to-digital Converter (unit 1), recycler (block 2), computing the fast Fourier transform (blocks 3, 9.1-M and 13), the complex multiplier products (blocks 4.1-M, 7, 10.1-M and 14), calculators reverse the th fast Fourier transform (units 6.1-M, 11.1-M and 16) and permanent storage devices (blocks 5, 8, and 15). However, this device [11] (prototype)that implements a method for detecting the complex of broadband frequency-modulated signals with filtering, scale-time domain [11], has the following disadvantage. Digital implementation of operations the calculation of the forward and reverse continuous (redundant discrete wavelet transform with arbitrary small discretization step of scaling factors αm=α0m, 1<α0<2, which analyzes the signal with arbitrary precision measuring scale, is not effective from the point of view of the required performance and the computational and hardware costs. So, for the numerical implementation of the forward and reverse CWP (the main components of the operation pre-filtering the useful signal to background noise in the large-time domain [11]) of the input sample length of N samples requires the following digital resources: - in memory ROM (block 5) should be stored M the calculated pre-compressed copies of the spectrum of the source wavelet - to implement the direct calculation of the wavelet transform with arbitrary discrete step scale in the spectral region required: one operation direct FFT (block 3) and M operations inverse FFT (units 6.1-M), or about - to implement the multiplication of the filter function H(m, n) and the wavelet transform of the input process Wx(m, n) requires MN complex multiplications (blocks 4.1-M); for implementation of computing the inverse wavelet transform with arbitrary discrete step scale in the spectral region required: M operations direct FFT (blocks 9.1-M) and inverse FFT operations (blocks 11.1-M), or, respectively, about MNlog2N complex multiplications; plus MN complex multiplications in the multiplier products (blocks 10.1-M); Total: - memory required ROM - 2 MN; - the required amount of computing Below is a new device that implements the method for detecting the complex of broadband frequency-modulated signals filtered scale-time domain, based on the use of calculators direct and inverse discrete wavelet transform (DWT) with a discretization step of scaling factors equal to 2, is much more effective from the point of view of performance, computing, and hardware cost than the prototype. Thus the immunity of the proposed new detector, as was shown by the simulation environment MathCadl3, almost completely coincides with the immunity of the device-prototype [11], i.e. has the same winning (11) relative to the classical matched filter. Theoretical foundations of the fast algorithm of discrete wavelet transform When the discrete wavelet transform the necessary discretization of the values of the parameters α and τ, while maintaining the ability to recover the signal from its wavelet transform should be performed as follows where m, n ∈ Z; α0>1; τ0≠0. The shift parameter depends on the parameter scale. With scale increases and the step size of the shift. At step shift τ0=1, the corresponding one odce is, the basis of wavelet functions is represented in the form Under certain, very specific, the choice of the wavelet ψ and initial values α0τ0a discrete set of functions ψατforms an orthonormal basis in the Hilbert space [5]. In particular, if the discrete EAP, without loss of generality, choose α0=2 and τ0=1 I.e. in digital form when fiberboard minimum step logarithmic scale compression option (or its base) is equal to two, and the minimum step of setting time shift - unit (i.e. one reference). For this method the discretization parameters compression and shear developed all known efficient algorithms fast discrete wavelet transform (DWT - discrete wavelet transform) [5-8]. In this case, compression of the original signal in the 2mtimes sufficient to produce an appropriate number of times thinning even-numbered or odd time samples. This method is implemented scaling of signals in the algorithms of fast discrete wavelet transform. In contrast to theory of continuous EAP in theory of discrete orthogonal EAP, or rather, in theory cranemaster analysis [5], in addition to the functions of a wavelet ψ(t) introduces another important concept is the so-called scaling function (or the scale of Bermuda function) φ(t). In the literature of the scaling function φ(t) is sometimes called the "father" wavelet by analogy with the mother wavelet ψ(t). Functions of the wavelet and scaling functions in theory of fiberboard are associated with discrete sequence gnand hn. Function ψ(t), φ(t) and the sequence gnhnlinked by the following key ratios When calculating the fiberboard is possible iterative calculation of the discrete wavelet coefficients cj,kand dj,kwithout the direct use of the paternal functions φ(t) and the mother wavelet ψ(t}. For an arbitrary phase decomposition of j can be written having thus fully discrete decomposition of the signal (direct fiberboard). Sequence hnand gnare essentially impulse response digital low - pass and high-pass filters. The coefficients of the approximation of cj,kand detail of the dj,k. have "half" length compared to cj-1,k. For a matrix description of the procedure fiberboard through the vector vjdenote the sequence of finite length cj,nfor some non scale j (or the step of calculating the DVP). At each stage fiberboard this vector is converted to a vector Vj+1which which contains the sequence c j+1,nand dj+1,neach of which is half-length. The transformation can be written in the form (18) matrix multiplication vj+1=Mjvjwhere Mj- a square matrix consisting of zeros and elements of hnmultiplied by As an illustration we can consider the following example. Take the filters gnand hnlength L=4, the sequence of signal c0length N=8, and as the initial value of the scale j=0. Then the operation of the matrix-vector multiplication can be represented in the form The sequence gncan be obtained from hnaccording to the formula: gn=(-1)nh-n+2L+1to rewrite (18) in the form Thus, the operation (19) is one step fiberboard. Full DVP algorithm is an iterative multiplication of the upper half of the vector vj+1for a square matrix Mj+1whose size is 2D-j. This procedure is repeated D times until the length of the vector will be equal to one reference. Complete decomposition of the original signal c0,nlength N=2Dthe wavelet coefficients dj,nrequire D similar matrix multiplications (for j=0, ..., D-1 stages fiberboard. It should be noted that the total number of elementary multiplications in the algorithm fiberboard depends on the length L of the selected sequences of the hn, gni.e. the type of the wavelet and its order. Typically, the most famous of the discrete wavelets (Haar, Daubechies - different orders of magnitude) the length L of the sequence hnsignificantly less than the length of N samples of the analyzed signal. The most economical in this sense is the Haar wavelet length just two count. The final result fiberboard contains N wavelet coefficients: N - 1 coefficients parts dj,nand one factor approximation of cJ,0. Usually the result of a one-dimensional fiberboard write one string of length N coefficients (in particular, in applications of MathCad and MathLab). For the considered example, the DVP will be [d1,0d1,1d1,2d1,3; d2,0d2,1; d3,0; c3,0]. The coefficient of the approximation of the last level cJ,0go for the start of the implementation of the iterative calculation of the inverse fiberboard. Reverse fiberboard can be described by multiplying vj+1return matrix In foreign literature is considered a fast algorithm for computing fiberboard associated with the work Small (S.Mallat) [5]. The fast algorithm is Alla discrete wavelet transform of the original signal of length N samples is implemented in D=log 2N stages and requires only 2NL multiplications. The essence of the proposed device The proposed device - detector complex broadband frequency-modulated signals filtered scale-temporal region-based discrete wavelet transform is shown in figure 2, where: unit 1 - analog-to-digital Converter (ADC); block 2 - recirculator; unit 3 - the evaluator discrete wavelet transform (DWT); unit 4 - permanent memory (ROM) 1; block 5 - multiplier; unit 6 - ROM 2; unit 7 - calculator reverse fiberboard; unit 8 - the transmitter 2 FFT; unit 9 - the complex multiplier; unit 10, ROM 3; block 11 - calculator inverse FFT; unit 12 - the transmitter unit square; block 13 - a threshold device; block 14 - control device. Thus, the detector complex broadband frequency-modulated signals filtered scale-temporal region-based discrete wavelet transform includes: analog-to-digital Converter (unit 1), the input of which is applied the input signal, the output of the analog-to-digital Converter connected to the input of the recirculator (block 2), the output of which is connected to the first input of the transmitter fiberboard (block 3), the output of which is connected to the first input of the multiplier (block 5), you are the od of which is connected to the first input of the transmitter reverse fiberboard (block 7), the output of which is connected to the input of transmitter fast Fourier transform (block 8), the output of which is connected to the first input of complex multiplier (block 9), the output of which is connected to the input of the transmitter inverse fast Fourier transform (block 11), the output of which is connected to the input of the transmitter unit square (block 12), the output of which is connected to the input of the threshold device (block 13), the output of which is an output device; the first persistent storage device (block 4), the output of which is connected with the second inputs of the transmitter fiberboard (block 3) and evaluator feedback fiberboard (block 7); the second persistent storage device (block 6), the output of which is connected with the second input of the multiplier (block 5); the third permanent storage device (block 10), the output of which is connected to a second input of the complex multiplier (block 9); control device; (block 14), the outputs of which are connected to control inputs of analog-to-digital Converter (unit 1), recycler (block 2), evaluator fiberboard (block 3), multiplier (block 5), calculator reverse fiberboard (block 7), the solver fast Fourier transform (block 8), the complex multiplier (block 9), calculator inverse fast Fourier transform (block 11) and permanent storage devices (blocks 4, 6 and 10). The principle of the device is the following. To the input device enters the implementation of the input process x(t), which is fed to the input of the ADC (block 1) with a sampling rate that satisfies the requirements of the sampling theorem: ADC output (block 1) discrete samples are sent to the input of the recirculator (block 2), which is formed and with each new count is updated to the current discrete sample x(n) of length N samples. The generated current discrete sampling of the input process x(n) arrives at the first input of the transmitter fiberboard (block 3), the output of which the discrete wavelet transform of the input process Wx(n) in the form of a one-dimensional array of size N samples is supplied to the first input of the multiplier (block 5). With outputs of the first ROM (block 4) reads the sequence h(n) of length L samples (corresponding to the selected basis discretely wavelets) and is supplied to the second inputs of the transmitter fiberboard (block 3) and evaluator feedback fiberboard (block 7). From the output of the second ROM (block 6) is read pre-calculated array filter function H(n) (scale-temporal plane) of size N times, and is supplied to the second input of the multiplier (block 5), which outputs a one-dimensional array of length N samples of the multiplication (i.e. filtering scale-temporal region) Wx1(n) arrives at the first input of the transmitter reverse fiberboard (the POC 7), since the output of which temporary implementation of the filtered scale-temporal process area x1(n) is fed to the input of the transmitter FFT (block 8), the output of which a comprehensive range From the third ROM (block 10) is read pre-calculated complex conjugate spectrum of the reference signal The control device (block 14) performs synchronization of work: a / d Converter (unit 1), recycler (block 2), evaluator fiberboard (block 3), multiplier (block 5), calculator reverse fiberboard (block 7), the solver fast Fourier transform (block 8), the complex multiplier (block 9), calculator inverse fast Fourier transform (block 11) and permanent storage devices (blocks 4, 6 and 10). Thus, h is emom device detector for digital implementation of the forward and reverse fiberboard (the main components of the operation pre-filtering the useful signal to background noise in the large-time domain input samples length N samples requires the following digital resources: - in memory ROM (block 5) should be stored impulse response h(n) of length L samples (corresponding to the selected basis of discrete wavelets); - in memory ROM (block 8) should be stored pre-calculated one-dimensional array of filter function H(n) (scale-time domain) of size N samples; for realizing the calculation of direct fiberboard required 2LN operations of multiplication (block 3); - to implement the multiplication of the filter function H(n) and the DVP input process Wx(n) requires N multiplications (block 5); - to implement the calculation of the inverse fiberboard required 2LN operations of multiplication (block 7). Total: - memory required ROM - N+L; - the required amount of computing - (4L+1)N complex multiplications. Other operations that implement a consistent filtering (blocks 8-13), completely similar to the device prototype. To clarify the essence of the operation of forming large-scale temporal filter function H(n), the filtering operation of the input process x(n) scale-time domain using fiberboard and illustrate the gain in noise immunity of the proposed detector signal of known form in relation to the classical matched filter figure 3-15 shows the simulation results (obtained in MathCad) of the processing, the chirp signal, discover the background of Gaussian white noise, classical coherent filter and the proposed detector. Figure 3 shows the reference chirp signal. Figure 4 shows the interference as Gaussian noise. Figure 5 shows the cumulative input process of the useful signal and the interference. Figure 6 shows the result of fiberboard reference chirp signal using discrete wavelet Daubechies 20 order). Figure 7 shows the result fiberboard total process of the useful signal and the interference. On Fig illustrates the kind of large-scale temporal filtering functions for the chirp signal. Figure 9 shows the result of filtering in scale-time domain (i.e. multiplication DVP input process on the filter function). Figure 10 shows a view of the filtered signal resulting from application of the reverse fiberboard. Figure 11 shows a view of the envelope of the autocorrelation function of the reference chirp signal. On Fig see the response of the detector on the basis of the classical matched filter. On Fig see the response of the proposed detector with additional pre-filtering on the basis of DVP and subsequent coherent filtering. On Fig shows the normalized response of the detector on the basis of the classical matched filter. On Fig shows the normalized off the to of the proposed detector with additional pre-filtering on the basis of DVP and subsequent coherent filtering. References 1. The Paul Burdick B.C. Analysis of hydroacoustic systems. Leningrad: Sudostroenie, 1988, 392 S. 2. Lezin US an Introduction to theory and technique of radio systems. M.: Radio and communication, 1986, 280 S. 3. Hellstrom K. Statistical theory of detection signals. M: Foreign literature, 1963, 430 S. 4. Van-Tris, Theory of detection, estimation, and modulation, vol. 1, M.: Owls. radio, 1972, S. 744, v.3, M: Owls. radio, 1977, 661 S. 5. Daubechies I. Ten lectures on wavelets. Izhevsk: center "Regular and chaotic dynamics", 2001, 464 S. 6. Lidia NM Wavelet analysis: basic theory and examples of application. The success of the physical Sciences. Volume 166, No. 11, 1996, s-1170. 7. I.M. Dremin, Ivanov O.V., V.A. Nechitailo Wavelets and their use. The success of the physical Sciences. Volume 171, No. 5, 2001, s-501. 8. Deacons VP Wavelets. From theory to practice. M: SALTY-R, 2002, 440 S. 9. Saprykin, VA, Small CENTURIES, Lopukhin RV Method and device for rapid computation of the discrete wavelet transform of a signal with an arbitrary discretization step of scaling factors. Patent of the Russian Federation No. 2246132 from 10.02.2005 priority from 09.01.2003. 10. Cohen, L., Time-frequency distributions: a Review. TIER, 1989, t, No. 10, p.72-120. 11. Saprykin, VA, Small VV Method and device for detection of complex broadband frequency-modulated signals filtered scale-time domain. The patent on izaberete is the development of the Russian Federation No. 2282209 from 20.08.2006 priority from 07.12.2004 (Prototype). 12. Rabiner, Bgold. Theory and application of digital signal processing. M.: Mir, 1978, 848 S. Device detection complex broadband frequency-modulated signals filtered scale-temporal region-based discrete wavelet transform, containing analog-to-digital Converter (1), the input of which is applied the input signal, the output of the analog-to-digital Converter connected to the input of the recirculator (2), characterized in that the output of the recirculator (2) connected to the first input of the transmitter discrete wavelet transform (3), the output of which is connected to the first input of the multiplier (5), the output of which is connected to the first input of the transmitter inverse discrete wavelet transform (7), the output of which is connected to the input of transmitter fast Fourier transform (8), the output of which is connected to the first input of complex multiplier (9), the output of which is connected to the input of the transmitter inverse fast Fourier transform (11), the output of which is connected to the input of the transmitter unit square (12), the output of which is connected to the input of the threshold device (13), the output of which is an output device; the first persistent storage device (4), the output of which is connected with the second inputs of the transmitter discrete wavelet transform (3) and evaluator inverse discrete Wei the years-transform (7); the second persistent storage device (6), the output of which is connected with the second input of the multiplier (5); the third permanent storage device (10), the output of which is connected to a second input of the complex multiplier (9); control device (14), the outputs of which are connected to control inputs of analog-to-digital Converter (1), recycler (2), computer fiberboard (3), multiplier (5), transmitter reverse fiberboard (7), solver fast Fourier transform (8), complex multiplier (9), calculator inverse fast conversion Fourier (11) and permanent storage devices (4, 6, 10).
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