Index of content:
Volume 128, Issue 6, December 2010
- ACOUSTIC SIGNAL PROCESSING 
Comparison between direct-sequence and multicarrier spread-spectrum acoustic communications in time-varying channels128(2010); http://dx.doi.org/10.1121/1.3500670View Description Hide Description
Underwater communication experiments have been conducted in the Norwegian Oslofjord. Two modulation schemes are compared in a 7-kHz frequency band on a 14-kHz center frequency. The first scheme is direct-sequence spread spectrum (DSSS), using a 7-chip spreading code to achieve a raw data rate of 1000 bps on a single carrier. The second scheme is multicarrier spread spectrum (MCSS) and accomplishes spreading by using seven subbands. The DSSS receiver equalizes on the chips prior to explicit symbol despreading, whereas MCSS features joint multiband equalization and despreading. Four channels are examined, from nearly static to overspread. In slowly varying channels, MCSS offers the best performance. DSSS has the best tracking potential for rapidly varying channels, where the challenge is to obtain reliable chip decisions before symbol despreading. The tracking potential can be realized to some extent by hypothesis-feedback equalization. It is further shown that adaptive equalizers are capable of code conversion, i.e., the DSSS receiver can demodulate the MCSS waveform, and vice versa. Neither receiver requires knowledge of the spreading code in order to despread the data.
128(2010); http://dx.doi.org/10.1121/1.3500669View Description Hide Description
It is often enough to localize environmental sources of noise from different directions in a plane. This can be accomplished with a circular microphone array, which can be designed to have practically the same resolution over 360°. The microphones can be suspended in free space or they can be mounted on a solid cylinder. This investigation examines and compares two techniques based on such arrays, the classical delay-and-sum beamforming and an alternative method called circular harmonics beamforming. The latter is based on decomposing the sound field into a series of circular harmonics. The performance of the two signal processing techniques is examined using computer simulations, and the results are validated experimentally.
128(2010); http://dx.doi.org/10.1121/1.3505121View Description Hide Description
This paper addresses the problem of field directionality mapping (FDM) or spatial spectrum estimation in dynamic environments with a maneuverable towed acoustic array. Array processing algorithms for towed arrays are typically designed assuming the array is straight, and are thus degraded during tow-ship maneuvers. In this paper, maneuvering the array is treated as a feature allowing for left and right disambiguation as well as improved resolution toward endfire. The Cramér-Rao lower bound is used to motivate the improvement in source localization which can be theoretically achieved by exploiting array maneuverability. Two methods for estimating time-varying field directionality with a maneuvering array are presented: (1) Maximum likelihood (ML) estimation solved using the expectation maximization algorithm and (2) a non-negative least squares (NNLS) approach. The NNLS method is designed to compute the field directionality from beamformed power outputs, while the ML algorithm uses raw sensor data. A multi-source simulation is used to illustrate both the proposed algorithms’ ability to suppress ambiguous towed array backlobes and resolve closely spaced interferers near endfire which pose challenges for conventional beamforming approaches especially during array maneuvers. Receiver operating characteristics are presented to evaluate the algorithms’ detection performance versus signal-to-noise ratio. The results indicate that both FDM algorithms offer the potential to provide superior detection performance when compared to conventional beamforming with a maneuverable array.
128(2010); http://dx.doi.org/10.1121/1.3504656View Description Hide Description
Near-field acoustic holography (NAH) is an effective tool for visualizing acoustic sources from pressure measurements made in the near-field of sources using a microphone array. The method involving the Fourier transform and some processing in the frequency-wavenumber domain is suitable for the study of stationary acoustic sources, providing an image of the spatial acoustic field for one frequency. When the behavior of acoustic sources fluctuates in time, NAH may not be used. Unlike time domain holography or transient method, the method proposed in the paper needs no transformation in the frequency domain or any assumption about local stationary properties. It is based on a time formulation of forward sound prediction or backward sound radiation in the time-wavenumber domain. The propagation is described by an analytic impulse response used to define a digital filter. The implementation of one filter in forward propagation and its inverse to recover the acoustic field on the source plane implies by simulations that real-time NAH is viable. Since a numerical filter is used rather than a Fourier transform of the time-signal, the emission on a point of the source may be rebuilt continuously and used for other post-processing applications.