Index of content:
Volume 126, Issue 5, November 2009
- ACOUSTIC SIGNAL PROCESSING 
Real-time calculation of a limiting form of the Renyi entropy applied to detection of subtle changes in scattering architecture126(2009); http://dx.doi.org/10.1121/1.3224714View Description Hide Description
Previously a new method for ultrasound signal characterization using entropy was reported, and it was demonstrated that in certain settings, further improvements in signal characterization could be obtained by generalizing to Renyi entropy-based signal characterization with values of near 2 (specifically ) [M. S. Hughes et al., J. Acoust. Soc. Am.125, 3141–3145 (2009)]. It was speculated that further improvements in sensitivity might be realized at the limit . At that time, such investigation was not feasible due to excessive computational time required to calculate near this limit. In this paper, an asymptotic expression for the limiting behavior of as is derived and used to present results analogous to those obtained with . Moreover, the limiting form is computable directly from the experimentally measured waveform by an algorithm that is suitable for real-time calculation and implementation.
Effect of reflected and refracted signals on coherent underwater acoustic communication: Results from the Kauai experiment (KauaiEx 2003)126(2009); http://dx.doi.org/10.1121/1.3212925View Description Hide Description
The performance of a communications equalizer is quantified in terms of the number of acoustic paths that are treated as usable signal. The analysis uses acoustical and oceanographic data collected off the Hawaiian Island of Kauai. Communication signals were measured on an eight-element vertical array at two different ranges, 1 and 2 km, and processed using an equalizer based on passive time-reversal signal processing. By estimating the Rayleigh parameter, it is shown that all paths reflected by the sea surface at both ranges undergo incoherent scattering. It is demonstrated that some of these incoherently scattered paths are still useful for coherent communications. At range of 1 km, optimal communications performance is achieved when six acoustic paths are retained and all paths with more than one reflection off the sea surface are rejected. Consistent with a model that ignores loss from near-surface bubbles, the performance improves by approximately 1.8 dB when increasing the number of retained paths from four to six. The four-path results though are more stable and require less frequent channel estimation. At range of 2 km, ray refraction is observed and communications performance is optimal when some paths with two sea-surface reflections are retained.
Forward propagation of time evolving acoustic pressure: Formulation and investigation of the impulse response in time-wavenumber domain126(2009); http://dx.doi.org/10.1121/1.3216916View Description Hide Description
The aim of this work is to continuously provide the acoustic pressure field radiated from nonstationary sources. From the acquisition in the nearfield of the sources of a planar acoustic field which fluctuates in time, the method gives instantaneous sound field with respect to time by convolving wavenumber spectra with impulse response and then inverse Fourier transforming into space for each time step. The quality of reconstruction depends on the impulse response which is composed of investigated parameters as transition frequency and propagation distance. Sampling frequency also affects errors of the practically discrete impulse response used for calculation. To avoid aliasing, the impulse response is low-pass filtered with Chebyshev or Kaiser–Bessel filter. Another approach to implement the impulse response consists of applying an inverse Fourier transform to the theoretical transfer function for propagation. To estimate the performance of each processing method, a simulation test involving several source monopoles driven by nonstationary signals is executed. Some indicators are proposed to assess the accuracy of the temporal signals predicted in a forward plane. The results show that the use of a Kaiser–Bessel filter numerically implemented or that of the inverse Fourier transform can provide the most accurate instantaneous acoustic signals.
126(2009); http://dx.doi.org/10.1121/1.3212919View Description Hide Description
An established model for the signal analysis performed by the human cochlea is the overcomplete gammatone filterbank. The high correlation of this signal model with human speech and environmental sounds [E. Smith and M. Lewicki, Nature (London)439, 978–982 (2006)], combined with the increased time-frequency resolution of sparse overcomplete signal models, makes the overcomplete gammatone signal model favorable for signal processing applications on natural sounds. In this paper a signal-theoretic analysis of overcomplete gammatone signal models using the theory of frames and performing bifrequency analyses is given. For the number of gammatone filters (2.4 filters per equivalent rectangular bandwidth), a near-perfect reconstruction can be achieved for the signal space of natural sounds. For signal processing applications like multi-rate coding, a signal-to-alias ratio can be used to derive decimation factors with minimal aliasing distortions.