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Sound projection is a method of using highly focused sources to project sound onto reflective surfaces to create the impression of surround sound, which can be realized via beamforming using loudspeaker arrays. Generally, these devices are mounted near a surface with a finite acoustic impedance, such as a TV accent wall. The beamforming performance degrades in this half-space with an impedance boundary when the filters are designed using transfer functions measured in the free field. This paper proposes a model for predicting the beamforming performance in a half-space with an impedance boundary in terms of the transfer functions in the free field. A mathematical model is presented, and the results are compared with practical measurements for a 16-driver line array in real-use environments. The results show that reflections from the boundary lead to "dark bands" in the beam, causing the beam to deviate from the designed direction. Using filters designed with transfer functions calculated by the proposed model yields improved beamforming performance. [ABSTRACT FROM AUTHOR]

The first aim of this paper is to give emphasis to the importance of assessing phase information when reconstructing and mapping a sound field. In fact, in acoustic analysis phase distribution is frequently simply either not considered or ignored, even though it can supply very useful information for the understanding of the mapping itself or for further analyses. In this paper a procedure to carry out phase mapping of acoustic sources in beamforming measurement is illustrated. The second aim of this paper is to propose a straightforward iterative optimization algorithm based on the monopole substitution starting from beamforming results. It allows for the reconstruction of sound field without the use of any matrix inversion. Both numerical and experimental validations of the method are presented. Results shown hereafter prove the effectiveness of the approach. [ABSTRACT FROM AUTHOR]

The coherence factor (CF) and Wiener postfilter methods have been proposed as effective approaches for reducing the output noise of the delay-and-sum (DAS) beamformer in ultrasound imaging. The theoretical framework between them was also established. However, past researches about the CF and Wiener postfilter methods mainly focused on the summation of an array signal. This paper analyzes the CF and Wiener postfilter in the synthetic aperture (SA) imaging mode, where two-dimensional echo data are recorded. Different CF definitions in the SA imaging are first given and the corresponding Wiener postfilter methods are then proposed, including a Wiener postfilter especially for the SA imaging, named as WienerSA. The performances of different CF and Wiener postfilter methods were evaluated on both simulated and experimental SA data. Results showed that the proposed WienerSA outperformed the other Wiener postfilters in reducing the sidelobe noise level. It obtained the highest contrast ratio among the Wiener postfilter methods, which was even higher than some of the CF methods. Meanwhile it could benefit a much higher contrast-to-noise ratio than those CF methods with further suppression of incoherent noises. Consequently, the WienerSA is believed to be a promising approach in enhancing the SA image quality. [ABSTRACT FROM AUTHOR]

A continuous wave signal received on a single hydrophone from a moving source is beamformed using the synthetic aperture created by the source, where the signal at each range is steered by a range-dependent phase, relative to the starting point. The range increment (aperture) is determined based on the Doppler shift estimated from the data, knowing the original signal frequency. Given a sufficient signal-to-noise ratio, the source depth can be estimated from the beam output, assuming knowledge of the mode depth functions based on the nominal sound speed and bottom profile in the area. The method is illustrated with simulated data and at-sea data. For real data, the signal phase contains a random, incoherent component caused by the (random) source motion and media fluctuations in addition to the deterministic range-dependent component due to source range change. A phase locked loop is introduced to remove the random component assuming that the random component fluctuates faster with time than the range-dependent phase. When a vertical array of receivers are available covering the depth span of interest, the beam output can be used directly to estimate the source depth. In this case, no knowledge of the acoustic environment is needed. [ABSTRACT FROM AUTHOR]