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1.
A blind estimator of the ocean acoustic channel impulse response envelope is presented. The signal model is characterized by a deterministic multipath channel excited by a highly nonstationary deterministic source signal. The time-frequency (TF) representation of the received signal allows for the separation between the channel and the source signal. The proposed estimator proceeds in two steps: First, the unstable initial arrivals allow for the estimation of the source signal instantaneous frequency (IF) by maximization of the radially Gaussian kernel distribution; then, the Wigner-Ville distribution (WV) is sequentially windowed and integrated, where the window is defined by the previously estimated IF. The integral gives the channel impulse response envelope, which turns to be an approximation to the blind conventional matched filter (MF). The blind channel estimator (CE) is applicable upon the following conditions: that the multipath channel contains at least one dominant arrival well separated from the others, and that the IF of the source signal is a one-to-one function. Results obtained on real data from the INternal TIde Measurements with Acoustic Tomography Experiments (INTIMATE'96), where the acoustic channel was driven by an linear frequency modulation signal, show that the channel's envelope detailed structure could be accurately and consistently recovered, with the correlation of the estimates ranging from 0.796 to 0.973, as compared to the MF result  相似文献   

2.
The paper discusses an inversion method that allows the rapid determination of in situ geoacoustic properties of the ocean bottom without resorting to large acoustic receiving apertures, synthetic or real. The method is based on broad-band waterborne measurements and modeling of the waveguide impulse response between a controlled source and a single hydrophone. Results from Yellow Shark '94 experiments in Mediterranean shallow waters using single elements of a vertical array are reviewed, inversion of the bottom parameters is performed with an objective function that includes the processing gain of a model-based matched filter (MBMF) receiver relative to the conventional matched filter. The MBMF reference signals incorporate waveguide Green's functions for known geometry and water column acoustic model and hypothesized bottom geoacoustic models. The experimental inversion results demonstrated that, even for complex environmental conditions, a single transmission of a broad-band (200-800 Hz) coded signal received at a single depth and a few hundred forward modeling runs were sufficient to correctly resolve the bottom features. These included the sound speed profile, attenuation, density, and thickness of the top clay sediment layer, and sound speed and attenuation of the silty clay bottom. Exhaustive parameter search proved unequivocally the low-ambiguity and high-resolution properties of the MBMF-derived objective. The single-hydrophone results compare well with those obtained under identical conditions from matched-field processing of multitone pressure fields sampled on the vertical array. Both of these results agree with expectations from geophysical ground truth. The MBMF has been applied successfully to a field of advanced drifting acoustic buoys on the Western Sicilian shelf, demonstrating the general applicability of the inversion method presented  相似文献   

3.
李焜  方世良 《海洋工程》2015,29(1):105-120
The conventional matched field processing (MFP) uses large vertical arrays to locate an underwater acoustic target. However, the use of large vertical arrays increases equipment and computational cost, and causes some problems such as element failures, and array tilting to degrade the localization performance. In this paper, the matched field localization method using two-hydrophone is proposed for underwater acoustic pulse signals with an unknown emitted signal waveform. Using the received signal of hydrophones and the ocean channel pulse response which can be calculated from an acoustic propagation model, the spectral matrix of the emitted signal for different source locations can be estimated by employing the method of frequency domain least squares. The resulting spectral matrix of the emitted signal for every grid region is then multiplied by the ocean channel frequency response matrix to generate the spectral matrix of replica signal. Finally, the matched field localization using two-hydrophone for underwater acoustic pulse signals of an unknown emitted signal waveform can be estimated by comparing the difference between the spectral matrixes of the received signal and the replica signal. The simulated results from a shallow water environment for broadband signals demonstrate the significant localization performance of the proposed method. In addition, the localization accuracy in five different cases are analyzed by the simulation trial, and the results show that the proposed method has a sharp peak and low sidelobes, overcoming the problem of high sidelobes in the conventional MFP due to lack of the number of elements.  相似文献   

4.
A model-based approach to solve a deep water ocean acoustic signal processing problem based on a state-space representation of the normal-mode propagation model is developed. The design of a model-based processor (MBP) for signal enhancement employing an array consisting of a large number of sensors for a deep ocean surveillance operation is discussed. The processor provides enhanced estimates of the measured pressure-field, modes, and residual (innovations) sequence indicating the performance or adequacy of the propagation model relative to the data. It is shown that due to the structure of the normal-mode model the state-space propagator is not only feasible for this large scale problem, but in fact, can be implemented by a set of decoupled parallel second-order processors, implying a real-time capability. In the paper we discuss the design and application of the processor to a realistic set of simulated pressure-field data developed from a set of experiments and sound speed parameters  相似文献   

5.
The Herault-Jutten network has been used to separate independent sound sources that have been linearly mixed. The problem of separating a mixture of several independent signals in free-field conditions or a signal and echoes in confined spaces is compounded by propagation time delays between the source(s) and the microphones because the conventional Herault-Jutten network cannot tolerate time delays. In this paper, we combine a symmetrically balanced beamforming array with the conventional Herault-Jutten network. The resulting system can adaptively separate signals that include delays introduced by the propagation medium. The proposed algorithm has been simulated in digital communication multipath channels where intersymbol interference exists. The simulation results show two clear advantages of the proposed method over the conventional adaptive equalization: (1) there is no penalty for very long impulse responses caused by long delays, and (2) no training signals are needed for equalization. The design of a multibeamformer to handle the source separation of multiple broad-band signals is also presented  相似文献   

6.
For cases in which a received signal is known exactly and the additive noise is white and Gaussian, the optimal detector can be implemented as a matched filter followed by a threshold comparator. However, the performance of this detector is sensitive to signal shifts and mismatch between the assumed and the actual structure of the received signal. As such, the use of a matched filter detector in a multipath environment can result in substantially poorer performance than expected. Here, it is shown that the use of the incoherent (or sliding) matched filter can also result in a substantial performance loss if the signal autocorrelation function is narrow relative to the interarrival times of the pulses. In contrast, a detector that compares the zero-zero lag of the matched filter cumulant sequence to a threshold has a performance that is relatively insensitive to multipath channels  相似文献   

7.
Active sonar detection in shallow water using the Page test   总被引:1,自引:0,他引:1  
The use of active sonar in shallow water results in received echoes that may be considerably spread in time compared to the resolution of the transmitted waveform. The duration and structure of the spreading and the time of occurrence of the received echo are unknown without accurate knowledge of the environment and a priori information on the location and reflection properties of the target. A sequential detector based on the Page test is proposed for the detection of time-spread active sonar echoes. The detector also provides estimates of the starting and stopping times of the received echo. This signal segmentation is crucial to allow further processing such as more accurate range and bearing localization, depth localization, or classification. The detector is designed to exploit the time spreading of the received echo and is tuned as a function of range to the expected signal-to-noise ratio (SNR) as determined by the transmitted signal power, transmission loss, approximate target strength, and the estimated noise background level. The theoretical false alarm and detection performance of the proposed detector, the standard Page test, and the conventional thresholded matched filter detector are compared as a function of range, echo duration, SNR, and the mismatch between the actual and assumed SNR. The proposed detector and the standard Page test are seen to perform better than the conventional thresholded matched filter detector as soon as the received echo is minimally spread in time. The use of the proposed detector and the standard Page test in active sonar is illustrated with reverberation data containing target-like echoes from geological features, where it was seen that the proposed detector was able to suppress reverberation generated false alarms that were detected by the standard Page test  相似文献   

8.
Effective communication and echolocation depends strongly upon the coherence of the channel through which the signal is propagated. Under certain conditions, the average coherence or equivalently, the spreading of a random channel may be described by a scattering function (SF). This represents a second order (energy) measure of the average delay, Doppler, and more generally, the spatial (azimuthal) spread that the signal experiences. The SF is analogous to the point spread function (PSF) discussed in the image processing literature and likewise describes the amount of “blurring” imposed upon the signal or scene transmitted. The SF will be briefly reviewed and its measurement by both direct (high resolution channel probing) and indirect (deconvolution) methods will be discussed. A new direct method using specially designed waveform pairs and a twin or uncertainty product (UP) receiver structure is introduced. Unlike high resolution matched filter implementations for direct probing that are limited by the fixed volume constraint of ambiguity functions, the UP receiver produces vanishing sidelobes and hence more nearly approximates a desirable two-dimensional delta characteristic. The improvement gained in SF measurement is illustrated by the results of an experiment in which the UP receiver and traditional matched filter implementations were used to directly probe an ocean multipath channel  相似文献   

9.
Estimation of Rapidly Time-Varying Sparse Channels   总被引:2,自引:0,他引:2  
The estimation of sparse shallow-water acoustic communication channels and the impact of estimation performance on the equalization of phase coherent communication signals are investigated. Given sufficiently wide transmission bandwidth, the impulse response of the shallow-water acoustic channel is often sparse as the multipath arrivals become resolvable. In the presence of significant surface waves, the multipath arrivals associated with surface scattering fluctuate rapidly over time, in the sense that the complex gain, the arrival time, and the Dopplers of each arrival all change dynamically. A sparse channel estimation technique is developed based on the delay-Doppler-spread function representation of the channel. The delay-Doppler-spread function may be considered as a first-order approximation to the rapidly time-varying channel in which each channel component is associated with Doppler shifts that are assumed constant over an averaging interval. The sparse structure of the delay-Doppler-spread function is then exploited by sequentially choosing the dominant components that minimize a least squares error. The advantage of this approach is that it captures both the channel structure as well as its dynamics without the need of explicit dynamic channel modeling. As the symbols are populated with the sample Dopplers, the increase in complexity depends on the channel Doppler spread and can be significant for a severely Doppler-spread channel. Comparison is made between nonsparse recursive least squares (RLS) channel estimation, sparse channel impulse response estimation, and estimation using the proposed approach. The results are demonstrated using experimental data. In training mode, the proposed approach shows a 3-dB reduction in signal prediction error. In decision-directed mode, it improves significantly the robustness of the performance of the channel-estimate-based equalizer against rapid channel fluctuations.  相似文献   

10.
This paper deals with the basic modeling problem in underwater acoustics that is the characterization of the channel between a transmitter and a receiver. The problem is analyzed here using an array of sensors that receive PSK signals emitted by several sources. Data come from an experiment realized by a physical system situated in the Mediterranean Sea. In order to identify the multipath channel, we need to access the propagation time delay and the angle of arrival of each propagation ray. However, many of these acoustic ray paths are too close to be separated by classic processing methods (matched filter, beamforming, etc.); new methods with better resolution must be applied in order to analyze the experimental signals and to determine their arrival time on the array of sensors. After a presentation of this problem, we will first discuss high-resolution methods that are usually applied in the localization problem; we will then focus on wavelet packet analysis which provides good results by improving the temporal resolution of acoustic signals  相似文献   

11.
Channel temporal variability, resulting from fluctuations in oceanographic parameters, is an important issue for reliable communications in shallow-water-long-range acoustic propagation. As part of an acoustic model validation exercise, audio-band acoustic data and oceanographic data were collected from shallow waters off the West Coast of Scotland. These data have been analyzed for temporal effects. The average impulse response for this channel has been compared with simulations using a fast broad-band normal-mode propagation model. In this paper, we also introduce a novel technique for estimating and removing the bistatic reverberation contribution from the data. As propagation models do not necessarily account for reverberation, it has to be extracted from the signals when comparing measured and modeled transmission loss  相似文献   

12.
In this paper, the effect of channel phase coherence upon a matched filter envelope detector output is investigated for a pulsed radar or active sonar. A novel model for the correlated channel phases allows the explicit calculation of the loss in detection performance using the deflection criteria. The theoretical model yields good agreement with simulations when the phase correlation coefficients between the first and last pulses are between 0.1 and 1.0. It is shown that a 3-dB loss in performance, as compared to the optimum detector for perfect coherence, requires phase correlation between adjacent pulses ofrho_{i,i+1} = 0.91, 0.96, and 0.96 for 10, 20, and 30 pulses, respectively. On the other hand, the same performance is obtained with a noncoherent combiner of the matched filter pulse returns when correlation between adjacent pulses,rho_{i,i+1} = 0.8, 0.835, and 0.84 for 10, 20, and 30 pulses, respectively. Ifrho_{i,i+1}is smaller than these quantities, one is better off performing noncoherent detection.  相似文献   

13.
Source localization using subspace estimation and spatial filtering   总被引:1,自引:0,他引:1  
Subspace-tracking algorithms have traditionally been unable to deal with a large number of sources and at the same timepreserve their computationally efficiency, since, typically, efficiency goes down as the cube of the signal subspace dimension. One solution to this problem, which is presented in this paper, is to use a newly developed algorithm for the design of spatial filters in matrix form, in order to spatially filter the incoming data snapshots. The result is that the signal subspaces are confined to small angular sectors and, thus, the effective number of signals present is reduced. A method is developed for designing spatial filters in an efficient manner by formulating the design procedure as a rank-deficient linear least-squares problem. The source-bearing estimation is done using the signal-covariance matrix, which is updated using a recently developed fast algorithm, which is necessary in situations where one or more sources are nonstationary. The combination of the subspace-based bearing-estimation and spatial filter algorithms is shown to give good performance in cases of medium signal-to-noise ratio and is capable of resolving sources that are below the resolution limit of both conventional and adaptive beamforming. In addition, the use of spatial filtering makes it possible to estimate bearings for more than N narrow-band sources, using an N-element array. An example illustrating this capability is given.  相似文献   

14.
In the analysis of crosshole tomography data, the first step is to estimate the arrival time and amplitude of the multi-path arrivals which comprise the received signal. Normally algorithms such as matched filter are used to determine the arrival times. However, when the bandwidth of the signal is small, this method cannot resolve closely spaced arrivals. We, therefore, investigate the performance of a simulated annealing algorithm in estimating the amplitude scaling factors and delay times of the separate arrivals in a signal composed of closely spaced arrivals with added noise. The algorithm is applied to field data collected during a crosshole tomography experiment conducted in sea ice  相似文献   

15.
A time domain synthetic reflection seismogram is detailed and, as a limiting condition on this model, the analytic reflection impulse response for a one-dimensional lossless acoustic medium with piecewise continuous acoustic impedance is obtained. This analytic impulse response solution, in the structure of a sum of terms by order of reflection, provides insight to some of the poorly understood aspects of acoustic reflections from stratified and smoothly varying media as may be encountered in shallow marine sediments and elsewhere. It offers as well an approach for the inverse problem of extracting acoustic impedance profiles from reflection response data, though other effects (such as wavefront spreading, dispersive and absorptive attenuation, and wavelet broadening attendant with pulse propagation through a medium) need to be accommodated.  相似文献   

16.
A model-based approach is developed to solve an adaptive ocean-acoustic signal-processing problem. Model-based signal processing is a well-defined methodology enabling the inclusion of propagation models, measurement models, and noise models into sophisticated processing algorithms. Here, we investigate the design of a so-called model-based identifier (MBID) for a general nonlinear state-space structure and apply it to a shallow water ocean-acoustic problem characterized by the normal-mode model. In this problem, we assume that the structure of the model is known and we show how this parameter-adaptive processor can be configured to jointly estimate both the modal functions and the horizontal wave numbers directly from the measured pressure-field and sound speed. We first design the model-based identifier using a model developed from a shallow-water ocean experiment and then apply it to a corresponding set of experimental data demonstrating the feasibility of this approach. It is also shown that one of the benefits of this adaptive approach is a solution to the so-called “mismatch” problem in matched-field processing (MFP)  相似文献   

17.
Mesoscale eddies constitute the most energetic component of the variability of ocean currents. An attempt has been made for the detection of oceanic mesoscale eddy signatures over the Southern Indian Oceanic (SIO) regions using the dynamic topography derived from TOPEX/POSEIDON (T/P) altimeter data, by the signal processing technique, called matched filtering. After applying all the ocean and atmospheric corrections, data of a complete cycle of T/P over SIO has been used for detection of eddy signatures. The geoid undulations are removed from the data of corrected sea surface height from T/P and the resulting dynamic topographic data are passed through a matched filter designed to detect a generic eddy signature of Gaussian signal embedded in noise. The filter is optimized to detect eddies with amplitude 20 to 30 cm and diameters roughly 100?250 km. Out of all the analyzed data of T/P orbits over SIO a few examples are presented for brevity. Qualitative verification of eddies is done with some independent T/P sea level anomaly data over the region. The analysis shows that the matched filtering technique is most suitable for monitoring eddy signatures along the subsatellite track instantly over the remote and most hostile regions of the southern global oceans.  相似文献   

18.
Reliable,with high data rate,acoustic communication in time-varying,multipath shallow water environment is a hot research topic recently.Passive time reversal communication has shown promising results in improvement of the system performance.In multiuser environment,the system performance is significantly degraded due to the interference among different users.Passive time reversal can reduce such interference by minimizing the cross-correlated version of channel impulse response among users,which can be realized by the well-separated users in depth.But this method also has its shortcomings,even with the absence of relative motion,the minimization sometimes may be impossible because of the time-varying environment.Therefore in order to avoid the limitation of minimizing the cross-correlated channel function,an approach of passive time reversal based on space-time block coding (STBC) is presented in this paper.In addition,a single channel equalizer is used as a post processing technique to reduce the residual symbol interference.Experimental results at 13 kHz with 2 kHz bandwidth demonstrate that this method has better performance to decrease bit error rate and improve signal to noise ratio,compared with passive time reversal alone or passive time reversal combined with equalization.  相似文献   

19.
The main effects accompanying the propagation of an initially narrow complex modulated light beam in sea water are studied. Sea water, with respect to modulation waves, is shown to be similar to a medium with frequency dispersion with respect to electromagnetic waves; in the medium, the effects of the time focusing of a wave packet are possible. Based on the self-similar small-angle solution to the equation of radiation transfer, we estimate spatial-temporal characteristics of pulse-signal amplitude modulated by a complex signal. We propose a scheme for constructing underwater LIDAR using a complex modulated illumination beam and the processing of a received echo signal. The processing allows the separation of the modulated component and its matched filtration. Values of possible delays and broadening of the separated echo signal caused by dispersion properties of sea water are estimated.  相似文献   

20.
A key research area in underwater acoustic (UWA) communication is the development of advanced modulation and detection schemes for improved performance and range-rate product. In this communication, we propose a variable-rate underwater data transmission system based on direct sequence spread spectrum (DSSS) and complementary code keying (CCK), particularly for shallow-water acoustic channels with severe multipath propagation. We provide a suboptimum receiver that consists of a bidirectional decision feedback equalizer (BiDFE) to cancel both postcursor and precursor intersymbol interference (ISI). We also develop iterative signal processing and time-reversal (TR) diversity processing to mitigate the effect of error propagation in BiDFE. We present performance analysis on bit error rate (BER) for different data rates. Our works show that this new variable-data-rate DSSS-CCK is a suitable candidate for UWA communications over varying channel conditions and distance.   相似文献   

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