Studies On Key Techniques Of Ocean Acoustic Information Acquisition And Transmission

In this dissertation, ocean acoustic information implies the information of moving soundsources (ships, submarines, autonomous vehicles, etc) and the information of sea ambient noise.This information is essentially important to study the underwater signal detection andcommunication, and even to study ocean climate or acoustic environment, both theoretically andexperimentally.This dissertation is intended to study the key techniques on ocean acoustic informationacquisition and transmission with a small size platform. Small size platforms, including anchoredand moving platform, have drawn extensive interest all over the world, since they are suitable fordata acquisition of large space-time span. The main difficulties of information acquisition andtransmission with a small platform lie in low frequency weak signal detection, bearing estimationof moving sound sources in complicated sea ambient noise background, and high speed digitaltransmission in multi-path underwater acoustic channels.The main topics studied in this dissertation focus on characteristics of ship-radiated noise andsea ambient noise, weak signal detection, detection with acoustic intensity array, bearing and sizeestimation of sound sources, multi-path time delay estimation, sparse equalization, fractionallyspaced equalization, and blind equalization of underwater acoustic communication channels. Thetheoretical basis involved in this dissertation include digital signal processing, signal detectionand estimation, vector acoustic intensity theory, adaptive signal processing, array andcommunication signal processing.The main contributions of this dissertation are as follows:1. A numerical method for simulating noise field of arbitrary spatial distribution and ofarbitrary array shape is presented. This method is used for the analysis and validation ofperformance of array design, signal detection and bearing estimation in this dissertation.2. The statistical detection performance of energy detector, zero-crossing detector and powerspectrum detector is analyzed based on the Neyman-Pearson criterion. Theoretical analysis andsimulations on ship-radiated noise measured at sea show that zero-crossing detector has goodperformance.3. A new method of signal detection for low frequency sound sources with vector acousticintensity array (VAIA) of limited aperture is proposed. The spatial-time gain for acoustic signaldetection in isotropic noise is derived, and the ROC (Receiver Operating Characteristic) inGaussian noise is obtained. Both theoretical analysis and experimental results show that acousticintensity detection is superior to sound pressure detection.4. A new method of two-dimensional DOA (Direction of Arrival) estimation for low frequencysound source with the non-typical VAIA of small size is proposed. The azimuth and pitch anglesof the sound source can be estimated with the 3 intensity components measured with this array.The estimation errors, including the finite difference approximation error (FDAE), theinstrumentation channel mismatch error (ICME), and error caused by ambient noise, are studiedsystematically. Experiments conducted in semi-anechoic chamber show that, after correcting theFDAE and ICME, the precision of DOA estimation in isotropic ambient noise is about 2°.5. A novel scheme is proposed for the bearing estimation of the separate sections of a volumetarget with a non-typical VAIA. The bearings of the midway section and the section between themidway and the stern of the volume target radiating low frequency noise is estimated with the vector acoustic intensity, while the bearing of the stem section radiating high frequency noise isestimated with the high precision adaptive FIR time delay estimation method. Experimentsconducted in a semi-anechoic chamber show that bearings of the 3 sections can be estimated withsatisfying precision and the 3 sections are distinguishable. This result provides a new approachfor ship size estimation at short distance.6. A hybrid algorithm for the high resolution multi-path time delay estimation is presented.First, the estimate of multi-path time delay is obtained by using eigenvalue decomposition withthe MODE algorithm, then the time delay and the corresponding gain estimate is refined with theNLS (nonlinear least square) estimator. Simulation on the linear frequency modulated (LFM)signal measured at sea is presented to prove the effectiveness of our method.7. For the purpose of reducing the complexity of adaptive delay filter, a polarity correlator isadopted to estimate the time delay, thus avoiding direct estimation of the mean square errorfunction needed by the previously proposed MDRL (Maximum Deviation from Regression Line)algorithm, therefore resulting in much lighter computational load at little cost in performance.Computer simulations on the simulated underwater acoustic channel and data processing ofchannel measured at sea prove that our method is effective.8. A modified adaptive decision feedback equalizer (DFE) is proposed to exploit fully thelong sparse channels in high-speed underwater acoustic digital communications. Based on therelationships between the optimal feedforward filter, feedback filter (FBF) on the one hand andthe channel impulse response on the other hand, influence of the precursor with different strengthon the FBF is analyzed. Our modified DFE only works on the significant taps predicted with theabove analysis. Simulations on sparse channel measured at sea confirm that our modified DFEhas the advantages of lighter computational load and faster convergence over the conventionalDFE with only a very small cost in performance.9. A Y/2 fractionally-spaced sparse complete feedback equalizer (CFE) scheme is proposed toavoid the performance deterioration caused by symbol timing error in the receiver. Theperformance of the scheme is analyzed based on minimum mean square error (MMSE) criterion,and the MMSE performance is proved to be the same as that of the DFE. The scheme alsoeffectively exploits the sparseness of the multi-path channels with large delay spread and reducesthe complexity of the equalizer with small performance degradation. Theoretical analysis andcomputer simulations over field measured wireless channel and underwater acoustic channelshow that the scheme performs better than symbol-spaced CFE and T/2 fractionally-spaced DFE.10. A new blind equalization method named ACMA (Absolute Constant Modulus Algorithm) isproposed. We give a fairly detailed theoretical analysis based on EPS (Error Performance Surface)that proves that our ACMA method has the advantage of much faster convergence rate. To furtherreduce the mean square error (MSE), the kurtosis of the output signal of the equalizer iscomputed and compared with a pre-selected threshold. If the kurtosis is smaller than thethreshold, which means the eye-diagram is open, the equalizer turns to the DFE mode, so as toachieve much lower noise floor. Both theoretical analysis and computer simulations showpreliminarily that our method has much better performance than CMA2-2.