An underwater broadband near-surface source depth estimation method based on the frequency domain interference structure characteristics of the deep ocean bottom bounce area is proposed. In this method, the approximate expression of the output amplitude spectrum of the vertical array received signal beam is derived by establishing the structure model of the arrival sound field in deep ocean bottom bounce area. By utilizing the periodic oscillation relationship between the amplitude spectrum and the source depth and the vertical angle of arrival, the received signal is mapped to the depth-vertical angle of arrival domain, which can be used to estimate the depth of broadband sources. The validity of the method is verified by simulation and analysis of influencing factors. The results of the South China Sea experiments show that the depth estimation results are in good agreement with the actual source depths by using the vertical short array with 64 m to receive the double explosive bomb signals with calibration depths of 50 m and 100 m respectively, and the estimation error does not exceed 7%, which verifies the effectiveness of the method.

A joint underwater acoustic localization method based on acoustic glider is proposed to verify the feasibility of multiple gliders for range and orientation estimation of underwater sources. Using the hydrologic and acoustic data acquired by gliders in the eastern Indian Ocean, the broadband pulse multiple-path propagation characteristic along the distance is analyzed. A range estimation method for sound sources based on pulse waveform matching using a single hydrophone is proposed. For a source with an unknown location, the structure of the pulse waveform can be obtained from the experiment. According to the environmental information acquired, the copy field for the structure of pulse waveform at the different ranges is numerically calculated. After the process of correlating the experimental and simulation signals, the range estimation is realized corresponding to the maximum value of the correlation coefficient. Based on this, range and orientation estimation of underwater sources are achieved through the collaboration of two underwater gliders. The results show that within 100 km, the range estimation can be achieved by the single acoustic glider and there are still some points of large estimation error. The collaboration of two underwater gliders is used to improve the accuracy of range estimation. For the source depth of 200 m, the root mean square error (RMSE) of range and orientation estimation is 2.5 km and 2.4° respectively.

A beam deconvolution technique for the location of the mixed far- and near-field sources is presented in this paper. The generalized two-dimensional convolution formalism inherent in the conventional beamforming (CBF) spatial spectrum yielded by the linear array observations for noncoherent mixed signals is derived. The Richardson-Lucy algorithm is exploited to focus beam power and obtain accurate spatial parameters of the near-field sources. The far-field components are extracted by mapping the mixed-source covariance matrix to the near-field manifold’s orthogonal complement space. The intrinsic one-dimensional convolution relationship is revealed. Hence, the angular domain beamforming deconvolution is used to estimate the far-field signals’ incident angles. The simulation results show that the proposed method improves the spatial resolution of the CBF spectrum and can separate the mixed sources after isolating the near-field components by projection mapping. Besides, a background noise level suppression of 10 dB for the far-field sources is achieved by comparing it with the existing schemes.

Traditional detection of broadband targets in passive sonars has low output signal-to-noise ratio and poor performance in a complex situation with multiple targets and strong interferences. To solve this problem, a target detection method is proposed based on the characteristics of the energy distribution of broadband signals in the frequency-wavenumber domain by using uniform linear array. The proposed method converts the array signal into the frequency-wavenumber domain and uses the characteristics of the width and the spatial distribution of the main lobes and the side lobes to discriminate the main lobes in the wavenumber domain. After discriminating the main lobes of the same target at different frequencies, the accumulation of main lobe energy and the number of main lobes are used as the azimuth spectra for target detection. The theoretical analysis and simulations show the proposed method only utilizes the main lobes which have prominent contributions to target detection, thereby reducing the influence of the side lobes dramatically and improving the detection performance significantly. The results of trial data processing show that the output signal-to-noise ratio of the proposed method can be increased by 5.58 dB compared to SPED and 8.73 dB compared to CED. In addition, the computing time is decreased by 46% compared to CED, which validates the superiority of the proposed method.

A method of three-dimensional beamforming, based on the multipole expansions of the sound field, is proposed in this paper and applied to the small aperture acoustic vector planar array. The desired three-dimensional beam pattern can be synthesized with a two-dimensional array structure by combining the vertical particle velocity component of the sound field and the relationship between the spherical harmonics and the multipole modes. Therefore, the required array structure can be simplified for a three-dimensional beamforming. For illustration, a 3 × 3 rectangular vector sensor array was considered. Simulation results indicate that the error introduced by the approximations for multipole modes extraction can be ignored and the obtained beam pattern approximates the theoretical beam pattern when the spacing of adjacent vector sensors is less than one-fifth of the signal wavelength. Compared to the three-dimensional beamforming based on the spherical harmonic decomposition of the sound field, the proposed method achieves comparable array gain with better robustness in the low frequency band. Therefore, a reliable approach for low-frequency three-dimensional target detection is provided.

To further improve the reliability and band efficiency of the underwater acoustic communication (UWC), this paper offers a polar code construction algorithm based on the Monte Carlo method with two improvements for polar coded modulation (PCM) communication needs and the underwater acoustic (UWA) channel characteristics to match the underwater acoustic polarized channel of coded modulation. These optimizations are optimizing Monte Carlo statistics parameters based on the dynamical cognitive of the channel, and jointing decoding and decision feedback channel estimation. The proposed polar code construction method is then verified by establishing a two-step PCM UWC application mechanism. Through simulation, the performance of polar code construction methods before and after improvement, and the robustness to time-varying channels are then compared and analyzed, as well as the performance of polar code modulated UWC system with joint bit-interleaved coded modulation (BICM) and combination with multilevel coded modulation (MLCM) under different mapping rules. Also, it is compared with the low density parity check (LDPC) coded modulation system. The lake experiment demonstrates that the PCM UWC scheme suggested in this paper effectively ensures the reliable transmission of information in the shallow water channel, achieves error-free transmission at a signal-to-noise ratio of about 14 dB and a communication distance of about 1 km, and also has better error correction than the LDPC coded modulation system under the same circumstances.

To deal with the inversion problem when the spatial structure of seafloor sediments is unknown, a trans-dimensional particle filter method is proposed in this paper, where the cross-spectral density of the pressure field is used to estimate the sediment layering structure and geoacoustic parameters. The simulation results show that the number of sediment layers and geoacoustic parameters can be effectively estimated using the proposed method, and the parallel particle calculation make this method more efficient than reversible jump Monte Carlo Markov chain (rjMCMC) sampling. A linear frequency-modulated signal received by a vertical line array in the South China Sea is processed using the proposed trans-dimensional particle filter. The inversion results of the sediment layers and geoacoustic parameters are similar to those acquired by rjMCMC inversion. The number of sediment layers and the posterior probability density of parameters can be effectively estimated using this method.

A rapid calibration method for marine geodetic control networks based on the combination of beacon mutual ranging information and absolute coordinate information is proposed. Compared with existing methods, the rapid calibration method reduces the initial condition restriction of free net adjustment; the number of absolute calibration points is reduced by more than 50%, and the calibration efficiency and absolute coordinate precision are significantly improved. In addition, several typical formations under ideal conditions are proposed by analyzing the formations and absolute calibration points. The shallow water test results show that the calibration efficiency is improved by 50% when the calibration precision of the reference network is greater than 0.2 m and the calibration precision of the reference network is greater than 0.2 m.

Due to the low power conversion efficiency of parametric array, a method proposed in this paper can be used to improve the power conversion efficiencies of parametric arrays by adjusting the specific acoustic impedance. The Kuznetsov equation that describes parametric arrays is first studied, and the influence of Lagrangian densities on difference-frequency waves is analyzed. And the relationship between the Lagrangian density and the acoustic impedance is obtained. Subsequently, the method of arranging perforated plates around the parametric array to adjust the acoustic impedance is proposed. The proposed method is validated through numerical simulations. The results show that this method can improve the power conversion efficiency of the parametric array by using perforated plates to change the acoustic impedance.

A method that uses a random surface to improve measurements of the radiated sound power in a non-anechoic tank is proposed. Based on the finite element method, the feasibility of improving sound power measurements at low frequencies by random surface in a non-anechoic pool is analyzed. The sound power of a spherical source is measured in a non-anechoic tank with a size of 1.2 m × 1.0 m × 0.8 m. Experimental results show that in contrast to a tank with a steady surface, once a random surface is generated using a water pump, the diffusion is significantly improved: (1) the Schroeder frequency is decreased from 10015 Hz to 8370 Hz, indicating the range of measurement is extended to lower frequencies; (2) the frequency response curves measured by spatially averaging techniques have smaller fluctuation and is closer to free-field value, that is, the accuracy of measurement is significantly improved. The proposed method contributes to obtaining more accurate narrowband measurements of underwater target acoustic characteristics in underwater environment environments.

When the sensitivity of a large-size vector hydrophone is calibrated in a standing wave tube, the distribution of the calibration sound field will be uneven, resulting in a large error in the calibration results. In the case that the accurate analytical expression of the calibration sound field cannot be obtained, the sound field simulation analysis model is established based on the actual standing wave tube calibration device. The sound pressure and acceleration values of vector hydrophone and reference scalar hydrophone are obtained by numerical calculation, and the sensitivity correction factor is derived by combining the sensitivity calibration formula and the reference value. After using the correction factor, the maximum absolute error between the test results and the reference value of vector hydrophones with different sizes is reduced to less than 2.0 dB. The results show that the numerical calculation method can effectively reduce the sensitivity calibration error of large-size vector hydrophones caused by the non-uniform sound field and expand the test object of the vector hydrophone standing wave tube calibration device.

An acoustic vector sensor is an important sensor for capturing sound field information. In various applications, a precise integrated 2D sensor is desired, but manufacturing remains challenging. A novel 2D thermal convection based acoustic vector sensor is introduced, which consists of crossed multiple wires. The central crossed wire acts as a heater and the four right-angle shape wires act as thermal detectors. Based on this design, the sensor provides higher sensitivity and a direct two-dimensional acoustic particle velocity measurement. The sensor is optimized by numerical simulation and then fabricated using MEMS process. Finally, experiments have demonstrated that the device has good sensitivity response and orthogonal characteristics.

To study the broadband noise characteristics of jet impingement on defector with different impact distances, an experimental study is carried out by using far-field microphones in a full anechoic chamber. Comprehensive data of the nozzle-plate distance (L) range between 30D (D is the diameter of nozzle exit) and 2D are obtained and the spectral characteristics are discussed. Reducing the impact distance, the results indicate: (1) The spectra increase significantly in all frequency bands, leading to an increment of the overall sound pressure level by 10–17 dB in the direction of polar angles (α) of α = 120° when L < 10D; (2) The noise energy of downstream directions transfers to lower frequency and the spectra of α = 30° tend to collapse at different impingement distances when L < 10D; (3) Since medium and high frequency impingement noise has a little effect on the sideline directions, the spectra of polar angle of 90° attenuate rapidly which form a steep peak when L < 10D. The quantitative analysis prove that the proportion of sound power of defector increases linearly with the decrease of L in the direction of strongest noise (α = 120°). Most noise are produced by deflector for impingement distance less than the length of potential core, such as 80% at L = 7D–5D. Moreover, the directivity of deflector noise is obvious. Impingement noise or trailing edge separation noise is dominated in different directions. Correspondingly, the low frequency and high frequency noise plays a dominant role in the upstream and downstream directions respectively.

The dissipative effect of corrugated perforated plate on acoustic wave is studied in two cases of plane wave vertical incidence and grazing incidence in this paper. The results show that the acoustic performance of the corrugated perforated plate in both cases is different from that of the flat perforated plate. Under the condition of vertical incidence, the equivalent perforation rate can make the sound absorption coefficients of the corrugated perforated plate and the flat perforated plate coincide in the range of medium and low frequencies, but the corrugated perforated plate will have a double peak phenomenon that is different from the flat perforated plate at high frequencies. Further study reveals that it is due to the continuous change of the depth of the back cavity caused by the shape of the corrugated plate. Under the condition of grazing incidence, the corrugated perforated plate and flat perforated plate cannot be replaced by the equivalent perforation rate. The height and length of the corrugated perforated plate have obvious effects on its acoustic performance. When the angle of the corrugated plate is constant (the tangent angle corresponding to the plate height and a quarter of the plate length), the corrugated perforated plate with the plate length less than 75 mm has similar acoustic performance.

In photoacoustic imaging, image reconstruction suffers from the assumption of an ideal point detector with omnidirectional response, resulting in a degradation in the resolution of the reconstructed image. An image reconstruction method that takes into account the impulse responses and directionality of the ultrasonic detector is proposed in this paper. A high-quality image representing the optical absorption distribution is reconstructed by iteratively solving the inverse problem of the forward model incorporating the detector directivity and impulse response. Results of simulation and phantom experiments show that the proposed method can significantly improve image resolution and contrast compared with traditional reconstruction methods and other reconstruction enhancement methods that do not or not fully account for detector characteristics.

A robust automatic speech recognition (ASR) method using consistency self-supervised learning (CSSL) is proposed. This method uses speech simulation to generate the speech with different acoustic environments, then uses the self-supervised learning to extract the speech representations and maximize the similarity between the representations of the simulated speech. So invariant speech representations can be extracted in different acoustic environments and the ASR performance can be improved. The proposed method is evaluated on the far-field dataset, CHiME-4, and the meeting dataset, AMI. With the help of the CSSL and appropriate pre-training pipeline, up to 30% relative word error rate can be achieved compared to the wav2vec2.0. This proves the CSSL can extract noise-invariant speech feathers and improve the ASR performance effectively.

An end-to-end time-domain music source separation method based on dual-dimension sequential attention combing the structural characters related to instrument sound sources and contents of song pattern is proposed to address the insufficient specificity when characterizing the instrument sound sources in music. First, characteristic basis functions are weighted with different attention based on two dimensions, namely time and characteristic channels, because of the significant regularity of the occurrence of different instrument sound sources in different parts of the song pattern. Second, a multi-resolution frequency factor is introduced into the loss function to measure the difference between separated sound sources and ideal ones from both time and frequency domains at the same time. As shown by the experimental results on the MUSDB18 dataset, the separation results of instrumental sound sources can be improved by giving special attention to both the time-domain song pattern structure features and discrete harmonic features of the sound sources. Compared with Demucs, the most advanced end-to-end time-domain music source separation method, the signal-to-noise ratio index of this method is improved by 0.40 dB, with particularly outstanding performances in the separation of drum and bass audio sources, whose signal-to-noise ratio index is improved by 0.13 dB and 0.60 dB, respectively. The specificity of the characterization of sound sources can be improved through sufficient use of multi-dimensional a priori knowledge, such as the semantic content and acoustic feature of sound sources, thus improving the degree of separability of sound sources.